dex Otaku

You can find out without uninstalling it. Go to the "codecs" page in the audio decoder config, and set MP# to "disabled". Whatever other codec is on your system will then be used by SS. If the problem persists, it's not being caused by FFDshow.

Is it still doing it with SS 4.0? And what specific version of FFDshow are you using? Also - are you repeatedly trying to transcode the same tracks? I had SS 4.0 cack due to an error of my own [specifically, not switching FFDshow to 16-bit output] while testing [the night it came out] .. SS left invalid links in its Db to nonexistent OMA tracks [the transcoded ones that failed] which prevented those MP3s from being transcoded until I manually deleted the links [in the track properties]. Doing a database "optimisation" may possibly be another way to fixed that [since it should remove the invalid entries].

Make sure [if you're using FFDshow with SS] that FFDshow [on the "Output" page] is set to use 16-bit output ONLY [by default, all bit-depth boxes are usually checked and this should work - but if you force it to use a higher bit-depth by deselecting 16-bit, it WILL NOT work with SS], and that the "Connect to" dropdown is set to "any filter" [since SS is neither a directsound driver nor a waveout device]. I can only tell you it wasn't doing that here.

I was referring to MP3 of near or the same bitrate [i.e. 128kbps in that example]. ATRAC3's artfifacting has a specific spectral character, as does MP3's artifacting [as done any specific encoding/decoding algorithm], regardless of the encoder used. Encoder efficiency does affect the amount of artifacting present, but the artifacting itself is pretty consistent.

I can't recall who it was that was talking about the RH10 editing bug, where if you split a track, then delete part of the track, the unit gags .. the problem here isn't really a bug. It's that you have to commit the first edit [i.e. press "STOP"] before doing another. It's likely that other units behave the same way, so - there's your workaround that isn't really a workaround.

Heh. Nice analogy. In my case, my dislike of LP2 came about because of listening to lot of tracks encoded in it. Hardware-encoded LP2 is not so bad, but software-encoded [i.e. SS or SB] still sounds far worse than even MP3 to my ears - very harsh, almost gritty or granular sound, almost like sand being blasted into my ears.

I've been putting this through its paces since last night, and writing up a lot of notes on usage, evident bugfixes, &c. Will post once I've finished ul/dl testing. Most obvious deficiency to note so far: The AAC module does not read tags at all. I've tried this with m4a tracks created by both itunes and foobar2000 [FAAC encoder] and the tags are completely ignored. This pretty much makes the AAC support useless, unless you want to spend hours retagging everything with SS [the CDDB lookup thing takes about 10 minutes to look up a single album from here, so I'd call that less than useful if you have hundreds or thousands of AAC tracks to re-tag]. The difference here is in the codec; atrac3plus > atrac3, or so goes the assumption. I find the software-encoded LP2 [132kbps] to still be unlistenable for most types of music with my ears [which are by no means X-Men ears]. Given the obvious differences in how encoding is done between the a3 and a3+ codecs, I have no doubt that 128kbps a3+ would sound a great deal better than LP2. Mind you, I still haven't tested that myself, so it's only an educated assumption.

DL'd and installed. This looks -very- promising. For basic editing this might already be better than Audacity [i'm not partial to Audacity's somewhat clunky interface]. Thanks, ozpeter.

Hulumau: Combining tracks can be done after uploading from your recorder. Select multiple tracks in your library, in the order you want them combined, then go to the EDIT menu and select "combine". This deals directly with the PCM or atrac3plus files used by the library. You can export the combined results as WAV as usual. Most nonlinear editors will allow dropping multiple tracks on their timelines [examples: audacity [free/open source], kristal audio engine [free], sony vegas, adobe audition, steinberg nuendo, cakewalk sonar, &c.]. Different editors behave different ways when you do, though. Audacity's default behaviour is to drop each file into a separate track, which isn't what you want. [can't find an option to change] Kristal AE - haven't got this installed anywhere here. Vegas - puts everything on a single track in the order you selected, and puts the file you actually dragged [when selecting multiple files] first. Adobe Audition - makes separate tracks [can't find an option to change] Nuendo - asks whether you want separate tracks or on one track ... As you can see, there isn't a consistent standard between editors for this particular function. I tend to use Vegas most often, so I was aware of its default behaviour and made the bad assumption that other NLEs would act similarly. It would be worth checking out Reaper, and maybe finding out what Kristal AE's behaviour is, in any case. It can be found here: http://www.kreatives.org/kristal/ In the short term, using SS's combine function might serve you best.

No, there is no way to turn off this option. However, yes - if you have a high enough background noise level, it will keep the auto t.mark from doing its job. It has to be pretty loud, though, and likely would only serve to ruin your recording. Possible solutions: * Remove the trackmarks maually on the unit. This will take a long time and a lot of button-pressing. Not really advisable for so many tracks. * Combine all the tracks using SS. <blockquote>Knowing it's a single recording, this is likely what I'd do - export the original [uncombined] tracks to WAV [as a backup], then combine them with SS [make sure they're sorted correctly *first*]. SS takes a long time to do this, so - what I've done in the past is combine the tracks in blocks making up 10-15 minutes each, then combine those blocks, &c. until you have one track. Don't be deceived and think SS is locked up while it's doing this, though - just let it do the work, it will eventually finish. When you've got it down to one track, export as WAV again and edit as you will. </blockquote> * Most nonlinear editors allow you to drop more than one file at a time onto their timeline. Select all in Windows Explorer, then drop them all into your editor. Most behave in such a way that all files will show up in series, in order, and bumped right up against each other. No combining required, though it makes the timeline look messy. This method requires the least effort and incurs no wait.

The title says it all. The FAQ is pretty outdated even in terms of SS 3.4 [which fixes many of the issues addressed in the advice in the FAQ], I know. Still - waiting for 4.0, will do testing with it, and update the FAQ.

I've been close-mic'ing [25-50cm] instruments and even amp cabinets for about 15 years with various condensors without distortion. Most stage and/or studio mics have pretty high max SPL ratings [-WAYYYYYY- louder than almost any acoustic instrument is capable of being played], especially if properly biased [i.e. 48V phantom] and jacked into even a half-decent preamp with lots of headroom. Even mediochre portable studio or FOH mixing gear have worked fine in literally every situation I've ever encountered. I didn't start getting preamp or mic distortion until I started using MD, but then - it runs off one battery, you have to expect there to be some sacrifices.

On something like this, I'd trust wikipedia's info as "most likely accurate." So yes.

http://en.wikipedia.org/wiki/Sata

Good stuff, niftydog. Not really. Or, well, maybe. You can find transformers suitable for making simple balanced->unbalanced interfaces at many surplus electronics stores. In Canada there a line of stores called Princess Auto, many of which will occasionally have 1:1, 1:2 and 1:5 audio transformers in stock [though I will not vouch for their quality. In the world of analogue, you definitely get what you pay for.]. Balanced vs. unbalanced: If you have to run a balanced mic any kind of [cable] distance to an unbalanced recorder, *use the balanced cabling* for the long part of the run, and a balanced->unbalanced interface of some kind right beside your recorder. Balanced connections = less signal loss and more noise rejection. That said, the type of cable included with the NT4 is an example of a single-cable, no-power-required solution to the problem of getting the signal from the mic to the recorder, assuming the recorder is very close to the mic. Any solution of better quality will require outboard equipment, and most will require their own power. Thoughts on the Rode NT4: the only one I've had the chance to play with [a rental] came with one of Rode's 5-pin to 3.5mm TRS cables [as well as 5-pin to dual 3-pin XLR]. The signal output from this mic [using a fresh 9V battery] was so hot that it easily overloaded the mic preamp in my RH10 with any sound louder than someone shouting within a couple of meters. Jacked into the line-in and used as drum overheads, levels would reach the -12dBfs mark, but YMMV. The same mic also overloaded the input preamps on a M-Audio Delta 1010LT regardless of what their input gain was set for [eventual solution: don't use the preamps on the 1010LT at all, they appear to be useless with any mic more sensitive than an SM57]. Related stuff: http://en.wikipedia.org/wiki/Balanced_audio (PDF document) Mackie's "Balanced Lines, Phantom Powering, Grounding, and Other Arcane Mysteries" which is worth reading despite many people's opinion about Mackie. Jensen Transformers Application Schematics. In particular, for the real DIYers, (PDF document) JT-MB-C "Real" Mic Inputs & Phantom Power for Sony DAT which is exactly the interface needed for stereo [dual XLR] balanced mics -> stereo unbalanced [3.5mm TRS] mic audio. Would be nice if they also had a PCB design to download. Side-note: whether the 5-pin to 3.5mm cable used with the NT4 is suitable for your use really depends on what you're recording. For loud sources, it should be just fine. For quiet sources [i.e. nature recordings], a real balanced interface is more likely to be necessary.

There is also the issue that these were purchased through Minidisc Canada, not Sound Professionals. MD-Canada no longer sells the TFB-2s, and I actually don't know what their warranty support was for them, but I sincerely doubt they'll cover any product that has obviously been torn apart by the user. I could always ask them, of course, but I don't expect much of a response.

The FAQ needs to be updated. You no longer have to erase the tracks from their source disc. That said, it appears that most of us who use HiMD don't use it as 1st generation storage only - record on the unit, upload, edit+backup using the computer, erase the original and reuse it. Here's how I at least do things: * make my recording on the unit * title the tracks on the unit using SS [the names carry across through SS and exporting to WAV] * upload the tracks into SS * convert the tracks the WAV * duplicate and convert the WAV tracks to WavPack or FLAC [lossless packing formats with no copy protection/DRM] and back them up [from this point on, this copy is the "unedited master"] * erase the original disc as it's superfluous at this point * edit the recording using a non-destructive nonlinear editor [such as Audacity [free], Crystal Audio Engine [free], Adobe Audition, Sony Vegas, Nuendo, ProTools, Cakewalk Sonar, &c.] * burn CDs, make MP3s, do whatever with the edited version without any concern about copy protection/DRM The basic premise here is to free the recordings of any proprietary format and the copy protection/DRM used by them. Sony's formats are more relaxed in this regard with the more recent versions of SS, but you're still locked into using their file format. Hence the immediate conversion to WAV, which has no DRM and is a widespread format capable of being used by literally all editors. Advice: don't do you editing in SS. Even the most basic NLEs [like the open-source Audacity] have more features and will serve you far better, enabling you to work much faster than one can with the very basic split/combine functions built into SS. Also - recent testing done by some of us here has shown that SS no longer has problems uploading recordings that have been edited on the recorder itself. If you have the time and the inclination, not to mention a reliable power source [do it while plugged in with the AC adapter], marking your tracks on the unit itself [and removing any extraneous trackmarks if necessary] is just fine. This can actually make editing easier if you mark your tracks intelligently, and when you get to dropping the eventual exported WAV files into an editor to work on your "final," you'll have a pretty easy guide to follow with your pre-split material [every track on the unit becomes its own WAV file]. Cheers.

They won't be able to upload digitally via USB with a netMD model, but they can copy to any other device [including a computer] via analogue means [a.k.a. they way everyone's been doing it since the dawn of electro-mechanical sound recording]. They could also copy digitally via SP/DIF using a deck with optical output, or upload via USB using an RH1. Total bollocks! For one thing, the first-generation recording already has one pass of lossy compression on it, so it's not even CD quality the instant it's first recorded. For another, look back one paragraph.

Hmm. You know, I never even thought of this [purchase was July 2004]. I just ripped the cannister out of the plastic mount to see if it was reusable, so in any case I've already voided the warranty.

A mic preamp. See: Hints on how to control your levels and make undistorted recordings / AGC, Manual levels, Battery Boxes, Attenuators, Preamps?!?!? in the Live Recording forum. In the short term, if you mic is self-powered [uses a battery of its own] or if it's a dynamic mic [doesn't need power], the line-in might suffice if the source you're recording is very loud.

I'm saying goodbye to my SP-TFB-2s as one mic element finally died on me. If anyone knows where to get matched CZ034E [or others that are the same size, about 2.5mm high * 4mm diameter it looks to be] mic elements, please let me know. The TFB-2's plastic mounting hooks are easily reusable. As for what killed the element that died - while on loan to another sound artist, one of the tiny black "screens" fell off [the one on the left element]. Ever since then [lacking anything even remotely durable to replace the screen with] I've had problems when there was any significant humidity. The element finally gave out completely last night - somewhat ironically, since my last-ditch effort with the mic was to pull the screen off the right element, cut it in half so there was enough to the hole on each, and re-applied the new pieces. But no, it's dead. This puts me out of location recording for an undetermined and possibly indefinite period of time. Yeehaw.

Nice work, Malcolm. Thanks for the additional info.

I don't think the relationship between manual gain vs. the setting itself [1-30] is in decibels, but I could be wrong. The difference in gain between low-sens and high-sens is an additional 20dB.

It's possible that this is a difference between gen1 and gen2; I didn't test my RH10, only the NH700. And it might still not be infinite hold - it just lasted a looooong time .. 20-25secs is still a long time in itself.

I finally got SS to demonstrate it's "transfer quantity" bug for me. Yeehaw. I think it's because of something else I'm running at the same time, so I'm trying to figure out which soft it is. With OMA [atrac3/atrac3plus] tracks, there is no problem. With MP3s, a maximum of 32 tracks can be dropped onto the HiMD at a time. This is odd, because I have done this [120+ tracks at once] before without incident. Hence thinking it's caused by something else that's running. If I figure out which soft is causing the problem, I'll post it here.