dex Otaku

Ah yes - I'd never seen that post, myself.

One might think that if Sony wanted to increase the spread of use of MD or Hi-MD, they might release English product brochures as well.

Tracks are written contiguously on the first pass. When tracks are deleted, the space taken by them is opened. This creates fragmentation when new tracks are written; continuous play requires the MD / Hi-MD system's read buffers in order to work. Fragmentation is one thing that makes the read head skip back and forth. Rearranging the track order on a disc will also cause the head to skip back and forth during reading. Random play has the same effect. A disc that has been written in-order using SS, and never rearranged, will [nominally] play continuously from beginning to end without having to seek back and forth at all. A disc on random play will be skipping back and forth to find each and every track. Add possible fragmentation to that and you've got a lot more seeking activity than a freshly written disc playing normally from beginning to end. And yes, seeking uses power.

You mean Hi-MD, right? netMDs and HiMD with MD-mode discs do not play digitally over USB. Directshow is likely what does it, not TR, though yes, this is quibbling. You need a deck with optical out and a sound adapter with optical in. Portables generally lack the optical output needed for this, and Hi-MD currently has only one offering [the ~$800 Onkyo midi system described in the announcements forum] with an optical output.

God. The consequences of posting while drunk. Thanks for at least seeming to not be offended. FLAC itself comes with a little program called "FLAC frontend". Most of the interfaces I've seen so far [only 3-4] seem to follow the basics presented in it. It appears to be made by someone named Speek; eir address is included in the about box as speek at myrealbox dot com; perhaps the frontend is a place to start.

Yes. More accurately, dynamic range compression.

While you're at it, how about FLAC options that actually resemble FLAC options, i.e. packing level from 1-8 and such? [A good place to compare with is the interface for FLAC that actually comes -with- FLAC; the options you currently have for the dll don't resemble the FLAC options in any other program I've used that uses FLAC.]

Since SS 3.0 I have only been using it to export, actually [sorry marc =\ ]. Reliability has increased several orders of magnitude as has accessibility [being able to do it by simply right-clicking the track in SS's library]. There's also the fact that the only alternate format I use with HiMDRenderer is FLAC, and the FLAC files it generates cause stream errors in Foobar2000 [my primary audio player]. [The files are still usable with anything other than fb2k] I still use HiMDRenderer when I want to export tracks that are not uploaded from HiMD. Also, if you're going to be recording with the optical input, HiMDRenderer is your only choice for exporting tracks.

It looks to me like the ipod thingy supports analogue i/o with the receiver, the R1 line appears to be for remote control purposes only [like syncro start, probably].

See http://forums.minidisc.org/index.php?showtopic=9582 Note that the t-board's post came after the ones here.

Yes, at least one person has reported trying [and failing] using linux disc image software.

Yes, it will work, many of us [including myself] are using HiMD for exactly this purpose. See here: http://forums.minidisc.org/index.php?showtopic=7436

Are there any car manufacturers that don't include at the very least an AM/FM radio, if not cassette and/or CD player any more? To the best of my knowledge, at least in North America, basically every car has come with a radio since some time in the 1960s or earlier. If your car doesn't have a radio - try the junkyard. You can get factory-fit radios out of wrecks for as low as $10 around here.

Hmm. It's actually kinda pretty.

The 'D' in any netMD's model number means Downloader. This means there are no recording inputs. It is for downloading via USB only. You will not be able to record from any live source with a 600D.

Unlike the film industry, the music industry has no set standards for what levels are to be used on any medium, including CD. The general guideline is to "load the medium nominally," meaning several things: * the dynamic range is compressed to fit between the noise floor and the peak levels of the medium [compression has basically always been used for all analogue tape formats, LP, &c. that had less dynamic range than CD - this raises the average level of the recording; one reason being that it's then far enough above the noise floor for the medium's weaknesses to not get annoying] * Peak volume is kept below levels that cause obvious distortion * EQ is applied to also make up for the deficiencies of the medium * in certain cases [as with cassette tape] noise reduction [as with Dolby B, C, and S] or compansion [dbx I and II] is also applied With CD EQ is generally not applied, though there are exceptions to every rule or guideline. Compression is applied depending on the type of recording and its intended use [radio mixes and singles are traditionally more compressed than albums, for instance]. Peak volume is, of course, kept under the brickwall limit of digital. The fact that CD is digital does not force any kind of standard on the mastering processing at all. In the 1980s and part of the 90s, CDs were mastered so that the original dynamic range of the recording, without any compensation for the medium [since it didn't really need it], was kept. Basically, master tapes would be digitised and then made into CDs. Since then the rage has been to use what we now call bit-pushing, which is really just dynamic range compression made a thousand times worse than it ever was in the age of analogue. The advantage to it is that it makes a CD seem a lot louder by making it so you can get more volume out of your player [turning the volume up seemed to be unacceptable for some reason]. The disadvantages are many, including severe listening fatigue, obvious distortion in the recording [if you rip and look at almost any recently pressed CD you'll find that most tracks will actually look like they're solidly clipping, i.e. distorting, from beginning to end], equipment damage [because the average level is so much higher, and people turn it up so it's loud, and damage their equipment, not to mention their hearing], and so on. Standards sometimes exist by engineer, by studio, or even by record label. The only standard that really applies, though, is to keep peak levels below digital maximum. The average levels of a recording will change depending on the total dynamic range of the recording, i.e. the range between it's loudest and quietest parts. Since the dynamic range of basically every recording is different, the average volume is different. The only way to make up for this is to normalise the recordings. This is best done by measuring the RMS [average] volume of a track and adjusting the overall volume of it, then compensating for any distortion by doing none other than dynamic range compression to it. iTunes auto-levelling is rather dismal, if I may say so, but this is less a reflection on Apple than it is on the rather large difference in dynamic range from recording to recording. It is extremely difficult to make a normaliser and compressor/limiter that will work with everything you put through it without causing severe distortion to one extreme or the other [i.e. either quieter or louder material]. They chose a middling route and it happens to work with most recent recordings, but anything older [and IMO, properly mastered rather than totally mutilated as most things are now] will still end up being too quiet. Conversely, the film industry has rather rigid standards that have developed since its inception to make sure that volume levels in a soundtrack are always within a consistent range, so that any playback system in any theatre can reproduce it properly and with relative consistency compared to other theatres. THX has helped this in many ways. The audio industry has no such standards, and the medium, be it digital or analogue, imposes no standards other than to fit the recording onto it so it's playable.

Any tracks that are contiguous, i.e. that run from one to the next and need to be gapless on the final CD/MD/Hi-MD - yes, I join them all together, edit them as a single piece, and add track marks later. In the end this ends up being a faster way to do things, and is 100% guaranteed to maintain gapless playback on the destination medium without resorting to crossfades or anything like that. My editing chain usually goes something like this: * Hi-MD -> Sonicstage * export unedited files as master backup * combine/split contiguous tracks with SS * export tracks as editing master [after this step SS is no longer involved in any way] * edit the tracks * create CD layout + burn CD * rip CD with EAC to WAV image with DAO cuesheet * convert WAV CD image to FLAC and edit cuesheet to reflect this * convert the source WAV files to FLAC * rip and tag from the CD [the image file actually] to MP3 and any other desired formats * copy all files to DVD-R The MP3s &c. at the end of the chain are separate tracks, and are not inherently gapless.

Yes, only once. Going by what happened when SS before v2.3 botched tracks, it gets changed -before- the upload, which is why an incomplete upload would botch the tracks on the disc itself. Correct. This is the entire idea behind DRM - put it on the medium, but make it uncopyable. Actually, you can't return recordings to the PC, either. Once you've transferred something to MD or Hi-MD, you can only erase the track.

Making a disc image does not work. Please try using the search feature. This has been gone over more than once already.

If I'm not mistaken, Type R = a version [improvement] of SP encoding Type S = a version [improvement] of MDLP decoding

The MZ-NH700 / MZ-NHF800 recording in real time in SP mode [using the most recent Type R codec] and are available for around $200CAD brand-new. This would give you a piece of new equipment that supports ATRAC SP [with the newest ATRAC revision] that is also MDLP, netMD, and HiMD-capable, as well as having optical, line, and microphone inputs like your RH10 already has. Incidentally, the R37 is a good SP-only unit. No type-R, but it has good accessable controls [including switches for some things rather than menu functions] and a metal case.

Regarding Total Recorder: no, there's no way to avoid this. TR is simply doing its job of recording; SonicStage is what is inserting the gaps. Until SS is changed in this regard, there's nothing to be done about it. However - if you're willing to accept possible dropped samples, there's an option in TR that tells it to drop zero-sections in its recording. There's no guarantee that the beginning/end of tracks will actually meet exactly when using this, but it's worth a try. A440 already pointed out the only potential solution to auto-trackmarking [using auto time mark]. Not that you'll be happy to hear this, but congratulations - you're the first person to report lost uploads with SS 3.0. As I never use the editing features on my recorder [NH700], btu always upload [and don't use TR any more because I haven't lost a single upload since using SS 2.2], the only thing I can suggest is that editing the tracks on the unit itself may be the cause of your problem. A440's experience seems to contradict this, though. Sony's warning about uploading PCM and editing it are absolute bollocks. They are there because most people don't realise the amount of data that makes up a PCM recording, and how much time it can take to copy these files [even from hard disc to hard disc] and edit them. Anyone with even a basic understanding of how these things work [most people don't], a computer fast enough to handle manipulating files that are in the 500MB range [most people now do], and the patience [most people lack this] can basically ignore those warnings. I regularly work with [audio] files that are up to 1GB in size - you just have to accept that it takes time to do things with that much data. In the end, it takes less time to edit a single 1GB file than it does to edit 20 50MB ones. I also use SS's combine/divide features regularly, for both short and long recordings, and have never had a single problem using it for such. [i usually combine all contiguous tracks into single files to maintain gapless playback post-editing.] As for SS crashing on your machine - what else do you have running? Any software for other USB hardware? Are any other devices plugged into the same USB controller as the RH10? Do you have a lot of other programs running in the background? [incidentally, I upload with fair regularity with heavy system load with no problems.] What version of Windows are you running? Servicepack level? What hardware do you have? Point being: the problem might be SS interacting poorly with something else you're running. I didn't actually know this, but then, the -only- thing I use SS for is putting MP3s on my HiMD, and uploading tracks. I wouldn't even consider using it for burning CDs. This is a well-known issue, and one that many of us have been griping about for some time. I agree with A440's response re: MS's response. Boilerplate. Sony are the ones you need to dog about this, if anyone.

My argument is that lossless-packed -recording on the unit- is impractical [at this point in time]. Not that lossless packing itself is impractical.

I think the bassy sound might just be from the environment you were in. Fiddling around with a decent parametric / paragraphic EQ in your editor, adding either rolloff or fairly wide but mild notch filter at around the room resonance [obvious freq's to try are in the 150-200Hz and 400-500Hz range] should help to fix this. I've only skimmed this thread so far, and will read more later, so I might get back to you on this again.

This is interesting .. it would basically be crossfading the tracks and moving the track mark to the nearest frame, much as gapless mp3 playback works with the relevant plugins in winamp, foobar2000, &c. I always edit audio as contiguous files, so I never run into the gapping problems. Ever. Just add track marks in Nero or CD Architect, and burn. It also makes it a lot easier to handle disc-wide edits like normalising, compression/limiting, channel conversion et al that usually get done to an entire recording, not just single tracks. But then, I'm rather fussy about these things.