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Everything posted by dex Otaku
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Put straight: any processing in the chain colours sound, especially a pass of lossy compression/decompression. Repeated D/A and A/D conversions are included in this. Straight from the deck to the computer is always best, counting on your sound card having decent A/D conversion. But yes, I digress at the same time - I skimmed the posts, missing your first point there.
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No, I meant that it seems you're not missing anything. i.e. I'm admitting my mistake.
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It seems not, Sharpo51.
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Where did you hear the rumour of compatibility with existing decks? Existing literature [linked from minidisc.org] lists the possible capacity of 64mm DWDD discs as up to 4.7GB, and mentions the possible 2nd-gen HiMD with 2GB discs, but nowhere is compatibility with existing units addressed.
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Two thoughts: 1 - bagpipes are -loud-. It's possible that you're overloading the mic preamp. Try recording with manual levels and with them turned all theor nearly all the way down. 2 - the MDs I've used [and the HiMD I own] have no filtration before their A/D converter - when you put a frequency -in- that is over 22.05kHz, it generally ends up converted as a harmonic that is equally below 22.05kHz as the original tone was above it [i.e. simplified though not absolutely correct - 30kHz gives an audible harmonic at 10kHz]. It's possible that your bagpipes are causing audible distortion because the mic / MD are picking up and attempting to record sounds that it produces -above- your range of hearing.
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If it has an internal 9V battery, it needs no external power supply. The hum comes from powering a mic directly from the MD/HiMD itself, while plugged in with an AC adapter. You should not get hum using the NT4, in other words.
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You get the hum when the MD is powering the mic. I wouldn't even try using something like an NT4 [which is made to work optimally with 48V phantom power but also works with P24 and P12 systems, and shows a rating for 9V usage as well] without an external power supply, which would bypass the hum problem altogether.
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#1 - you're in the wrong forum. This forum is for issues related to live recording, and you're not inquiring about those. #2 - if you looked in the correct forum, for MD/NetMD, you'd find a stickied post containing the answers to your questions. Correction: it appears that sticky may have been one of the lost posts. Either that or I'm delusional, which wouldn't be altogether surprising.
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This is sort of irrelevant. Since lower-bitrate atrac/3/plus modes use joint stereo for recording [read: MS-stereo], if both channels are identical [a mono signal sent to both left and right] you are basically recording with twice the quality that you'd get with a stereo signal, since you're basically recording at full or near-full bitrate on the M channel with nothing [or next to it] on the S channel. It's a sacrifice that you can't record in mono and get twice the time out of the disc, sure. But given the option between twice the time or twice the quality, I'd go with quality, myself.
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Check out: http://www.dpamicrophones.com/ And click on "microphone university" They have some of the best info I've found on the net.
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Things that will slow down your transfers: * having lots of tracks on the disc [i.e. 100 tracks of 1 minute each will take WAY longer than 1 track of 100 minutes] * [possibly] having any other devices on the same USB interface as the HiMD * a busy system in general * using MD80s rather than HiMDs [HiMD discs, being higher density, both read and write at higher rates than MDs, even in HiMD mode] and, as they say - mileage may vary.
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Why, if your purpose is archival, would you go through the realtime process of copying cassette to MD then again from MD to your computer? Why not just plug the cassette deck into the computer, skipping the extra time spent, and skipping the one generation of lossy encoding?
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If you download one of the installer versions of the XVID codec they include the directshow filter and will be usable by WiMP after installation.
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SS's ID3 importing is rather dismal. The best solution I've found yet is to retain my rather old habit of being extremely anal-retentive about all files being tagged properly with both V1 and V2 tags. MP3BookHelper is the most versatile tagger I've found yet [after using many since about 1998] and it's open source [i.e. free]. See http://mp3bookhelper.sourceforge.net to download it. Among important things to note: * Make sure that both the V1 and V2 tags are present. SS seems to randomly pick which one gets used [which sometimes means truncated titles]. * Make sure track numbers are simply "1" or "01" and not "01/12" for example with V2 tags. The "/12" messes up SS and it will not pick up the track numbers properly, even if V1 tags are present with the simple "1" track number included. * If you're using Sony's mp3 converter tool, putting the tracks you want to import into a folder named with the album title ensures that SS imports the files into an album with that name, rather than gobbledygook or simply "imported files". And no, you can't digitally transfer standard MDs. At least - not without a home deck with optical output. You would be better off recording the tracks on your PC by analogue means than going through a second pass of encoding with your HiMD.
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Have you tried using the format option on the recorder itself?
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Sony's video cameras and other devices with "plug in power" for mics are also very well-known for this "feature". [i've also experienced this with Canon miniDV equipment. It appears that power filtration isn't a concern for equipment manufacturers.] It should only happen with powered mics though. I've met budget video crews who carry around their own power along with their equipment - marine or motorcycle lead-acid batteries in cases, and a 12V recharger, because getting good AC at any given location is iffy for one - secondly because of problems with powering condensor mics.
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I have noticed that with my NH700, regardless of how clean the power source is or whether the chokes are installed or not, trying to record with a powered mic while using the AC adapter is an exercise is utter futility. HUMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMM. On the other hand, with a no-name AA battery and a HiMD disc I have successfully made recordings in excess of 7 hours in length [with a powered mic, in Hi-SP mode] and still had enough juice to use the unit as a player for a couple of days [of heavy use] afterwards. PCM mode would be another matter, though.
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Digital recording from SACD or DVD-A
dex Otaku replied to watcher666's topic in Technical, Tips, and Tricks
DVD-A contains multiple formats in its standard, including [this is from memory so I might make errors here]: * AC-3 / A52 [Dolby Digital] from 2.0 to 5.1 channel * DTS [which counts as a PCM bitstream according to the spec] * PCM from 44.1kHz to 192Khz sampling rates, 16-24 bit quantisation, with channels limited by total bitstream bandwidth [higher sampling rate, higher bit res, fewer channels - lower sampling, lower bit res, more channels] as well as * Meridian lossless packing [MLP] in multiple sampling rate/bit res/channel configs MLP is the format to cause the most real excitement, in my opinion. Please note that the potential length of a given disc depends completely on the format used as well as the resolutions; higher res = shorter recording length, &c. -
On one hand, there is no reliable answer to this question. On another, there are no a/d conversions if you connect everything properly. The reason there's no reliable answer, though, is that directsound processes almost everything going through your puter audiowise. Disclaimer: this depends on your sound card and [their] drivers. Some drivers will allow completely unprocessed audio to flow from the source to the output; others, regardless of whether your choose to force digital out or not, thoroughly munge the data by resampling, requantising, &c. The M-Audio Revo 7.1 is a fine example of this: it's impossible [by my experience] to play a DTS audio CD through its coax out and have it decode properly with outboard equipment, for example. [Whatever you try to play gets reframed as 44.1kHz CD audio. Don't ask how many players I've tried this with.] The end result is that you will never really know if you're getting bit-for-bit copies. Which isn't really relevant, actually, for two reasons: first, the digital in on all MDs and HiMDs resamples the stream, even if it's already 44.1kHz. Second, if you're recording in atrac/3/plus on the second unit, the 2nd generation of encoding will introduce far more artifacts than proper resampling should. Another thing to note is that by playing the audio straight from HiMD using SS, SS is doing the atrac/3/plus decoding. It has been argued that SS's decoder is not as accurate as the hardware ones built into MDs/HiMDs, though no one has proved this asof yet.
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I'm kind of with aeriyn on this, except for on one point - a hard disc player is not permanent storage, and should never be considered permanent storage. If what you really want is to back up your LPs, I'd suggest using CDR or DVD+/-R. Example: you could make 24-bit/96kHz recordings of your LPs, compress them losslessly to FLAC or the format of your choice, write several albums to DVD+/-R, and use those files at any point you wished to create mp3s, AACs, WMAs, audio CDs, MDs, HiMDs, or if you happen to have a cutting lathe, even new LPs. If archival is your goal, uncompressed or losslessly-compressed audio is definitely the way to go. To MD's credit, it's the most durable portable removable media I can think of that isn't solid state. [How's that for vague?]
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If the backup was made using SS's backup utility, yes. If the backup was made by copying the SS library from one drive to another, no. Sorry, it's just that you were ambiguous about whether it was an actual backup or not. Cheers.
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Welcome to the forum, by the way.
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I'm with linzq on this: if you have a decent sound card in your computer, it's better to transcribe your LPs there first, do any editing that might be needed [clipping beginnings and ends, for instance], then convert them to whatever formats you prefer. Once it's on your computer, you can write CDs, make mp3s or FLAC or OGG or whatever other compressed files you like, as well as import the tracks to SonicStage to put them on HiMD. Incidentally, high-quality USB sound adapters can be had for less than $100USD that would blow the performance of $5,000USD recorders of 10 years ago right out of the water. Also: loss of quality after transferring from HiMD -> computer depends on what format you originally recorded in. If you plan to record in a compressed mode [HiSP most likely] then you will be incurring multiple generation loss by decoding to WAV on your computer, then re-encoding to another format. I do not recommend this. The recording quality of all HiMD units with analogue inputs should be equivalent, as they all have the same DSP. There might be minor differences such as what input buffer/amps are used, sure. I have the lowest-end recorder [NH700] with both mic and line inputs and have found that it consistently surpasses my expectations in terms of recording quality.
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The "rule" I mentioned is in fact a throwback to the analogue days, which I stick to because it tends to avoid exactly the kind of problems that people are experiencing with the EQ on HiMDs. The idea is that adding gain means adding noise. For all intents and purposes, the ideal type of equaliser is in fact complletely passive - without amplification at all. Another stage of amplification means another stage of both added noise and distortion. In a studio situation this tends to be important because various signals might be routed through multiple processors, cumulatively adding layers of noise and distortion for every one until it becomes noticeable. In the digital realm this becomes arguably somewhere between "less of a problem" and "a complete non-issue" depending on the processing techniques used, but I still stick with it because it's simply my rule of thumb, and with anything from home receivers to HiMD units to iTunes and Winamp still tends to avoid certain problems [such as channel clipping within amplified bands, and soft clipping or limiting]. So yeah. It's not that I'm old, really, but it is that I learned a few of the old ways first, which I tend to think of as an advantage. My favourite way to edit is still with 1/4" 1/2-track open reel tape, for instance. Mind you, what I actually use is a combination of different digital editors, each of which I find best-suited for certain purposes. Maybe I'm just a stick in the mud. Heh. As far as tradeoffs go, as long as the gain you apply to any specific band is less than would cause clipping in that band for whatever material you're listening to, there shouldn't be any problem. In practise this doesn't work well, though, especially with recently-mastered recordings, much of which are run straight into clipping a fair majority of the time before any post-processing is applied [i.e. as they come straight off the original disc]. I only use attenuation with my NH700 to correct for the rather dismal response curve of my Koss pseudo-canalphones, which ends up working quite well [surprisingly]. They're also of rather high sensitivity, so the volume when I'm listening rarely ever goes over 12-13/30 [as it becomes ear-splittingly loud above that]. As a result, I'm not likely to run into the soft limiting problem to begin with. I'd suggest that finding a better pair of ear/canalphones would improve everyone's listening experience far more than learning how to apply [a rather limited] EQ "properly" [it's all subjective, hence the quotes]. Attenuation isn't necessarily non-destructive, either. If anything, by definition attenuation is specifically destructive. In the end the idea is to do as little processing as possible, because all processing is by nature destructive in the sense that it's adding distortion of one kind or another.
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It's the same situation. The only place you can use the library is with either: 1) the installation of SS it was created with, or 2) any installation of SS, if restored from a backup created by SS on the originating machine [in which case it eradicates the library already on the machine being restored to, as well] (2) is assuming you had a SS-created backup, of course, which you don't.