dex Otaku

Any other comments from those who have listened? Constructive criticism is welcome.

See here: http://www.edirol.com/products/info/r1.html Note that media is mucuh more costly for this, though it's higher resolution and free of mechanical parts. You can also upload recordings totally in the clear.

Tape heads do not transduce at line level. Phono cartridges do not transduce at line level. They also do not reproduce all of their bandwidth ina linear fashion. Or, another way to look at it: they do reproduce in a linear fashion is compensation is added to the signal [pre-emphasis] before applying it to the media, and removed [de-emphasis] after transduced back to an electrical signal. This is why tape has different EQ positions for different tape formulations [affecting the EQ and bias freq when recording and EQ during playback]. It's also why phono preamps have standards determining pre- and de-emphasis curves so response comes out "correctly" i.e. the desired bandwidth is flat [-tish] from bottom to top. It is the same priniple that applies to a microphone. Analogue devices generally use a transducer to "read" the music; a microphone is a transducer, same as a tape head, same as a coil or crystal in a phono cartridge. The signals induced by any of them are miniscule, in millivolts. A preamp is used to amplify that to a managable level to make longer cable runs more practical [feet instead of inches] and noise rejection better, amon' other things. Some devices also have pre- or de-emphasis in their preamps. Microphones generally don't, but many have selectable cut or rolloff filters, especially if they also contain other electronics of their own [such as internal power in the case of a condenser]. Most gear that use line-level don't actually go through preamps. In this case, it's a piece of gear that is almost a misnomer; they usually have buffering preamps in them [which add noise but are great as components in, say, an electronic source-switch that you can run from a cheap IR remote control]. Higher-end units like this are simply passive switches that do not signal processing except for attenuation in the main "volume control". So. In the case of stereo equipment, a preamp generally is nothing more than a glorified switcher. Of course, it would probably also have a phono preamp in it, legitimising the name.

Just remember that what sounds good on those headphones, when you're probably squeezing them to your head to block the outside noise, is NOT what will sound good as a recording. If you know how to mentally compensate for the 'phones misgivings in that situation then you can at least approixmate something that will sound good. It does take practise.

As someone who has done editing both in the digital and analogue realms.. I miss 1/4" tape. There was something about shuttling 15ips tape to mark and cut your edits, the spools of annoying white splicing tape.. the sound of large solid-state headphone preamps driving actual watts straight to your ears, shwoooosh-shwooosh... Um, yeah. Honestly, I'm glad I learned how to do most things the "old way", with taped fly-ins and splice-editing and .. and .. Technically, I edited on digital, first. But I never did anything real with either until college, with 1/4" tape machines, it was so much fun. It's such a physical process, there's just something about.. about having control over something real.

You might try moving the mic farther away, and using the mic preamp with sensitivity set to low, and manual record levels. Mic farther from subject = lower level from the mic; no preamp overloads from blasting horns. Though admittedly, I don't think you would be overloading that mic anyway. Seriously, though. If that's it, then back off with the mic, problem solved. Why buy a preamp when you can spend $0 on moving the mic back half a metre.

Simple advice: be direct, and tell them you're interested ni the position, and that they could hire someone new in your old one. At the very least, put together an updated resumé and then hand in in to them as applying for that position. If you show interest, they're likely to notice.

The font for messages is slightly small for such a light colour on a lighter background. Simply changing the text to black would go a long way in terms of readability. Note that I'm using a calibrated 19" display at 1600*1200; improper [i.e. old-style] font sizes in html are much more noticeable like this. Proper styles [using points rather than pixels] work great, though.

Unity means no gain and no attentuation. If you plug a calibrated line-level signal into the HiMD, a manual level of between 18 and 19/30 gives you measurable unity gain. Mine is off by about -1dB at 18/30. The idea is that the HiMD should not be acting as either an amp or an attenuator when using it this way. The mic preamp should be given full resposibility as to how levels are set [i.e. use the preamp to do it]. Not all mic preamps have gain controls, so there are obvious exceptions to my opinion on this. edit: Note that there are different ways of measuring "unity" gain when digital come into play. What I'm calling unity here is when 0dBVU = 0dBfs; that's when a 0dB volume measurement - of the analogue signal - is equal to the highest level digitally recordable, 0dB full-scale. There are other ways of doing this. I assume that Sony's convention with consumer electroncis is that -12dBVU = 0dB, sort of like how cassette decks are set with 0dBVU at -10dBV. I tend to ignore the 'peak dash' in the MD's volume meter as anything other than the '-12dBfs mark,' though. Um. Yeah.

I find the best way to run any equipment is to do as close to actual calibration as possible. This is a bit fuzzy with mic preamps, [since there is no unity, just a "standard" amount of gain] though. "Unity" on the HiMDs is at about 18/30. If the gain control on your behringer has a centre position, check what its level is on the HiMD when it [the HiMD] is set at unity. Put the mic at a comfortable distance from the source, check the level, then adjust the preamp accordingly until you're comfortable with what's on the HiMD's meters. edit: Another point to press, one which I've made before - Even with a lowly 16-bit, 44.1kHz recorder, the quality is sufficient that you can leave yourself LOADS of headroom [i.e. 20-30dB] and still make a very high-quality recording. Leaving lots of headroom means never worrying about peaking at 0dBfs; you'll likely never get clipping. Most professional recordists I've seen run with their peak meters riding around -20dBfs. I also do this, unless the situation where I'm recording is predictably loud [like a punk concert] and you know the levels will never get above what yougre already hearing because this is as loud as anything near you can get.. In other words, unless you can predict that what you're setting you peaks at is actually the 'high average' of -everything- you want to record, don't be afraid to crank down that mic gain so that you have loads of headroom to protect yourself from digital clipping. After all, it's easier to raise the overall volume and apply gentle compression or limiting to the really loud parts later than it is to try to resurrect something useful from a bunch of distorted, clipped garbage. Running hot is not the best way to do things. If you know you'll be doing at least one pass of editing later, leave yourself a broad margin of safety.

The key here is "to minimise noise." Chances are, plugged straight into the mic preamp, the ambient noise level of wherever you are recording will be much higher than the self-noise of the mic or the preamp's own noisefloor. If you're recording *loud* sounds, go ahead and use the line in. Otherwise, the mic preamp gives you more control over things. Also, preamps are more than feasible in the field. Take a look on sites like http://www.minidisc-canada.com for an idea of the preamps that are available. Most are roughly the same size as an MD, run on a 9V battery [meaning -much- higher headroom before the preamp goes into clipping], and also act as a battery box should your mic need "plug-in power."

Yes: use a better microphone. Seriously though, the placement of the mic is probably the most crucial element there is to gettinga balanced-sounding recording. If at all possible, monitor the mic with headphones that have have some degree of isolation [even wearing earbuds with construction-type hearing protection over top works, I've done it even if it looks odd] and move the mic around to figure out where it sounds best. This does, of course, require a fair amount of knowledge as to what the deficiencies of your 'phones are, to avoid making it just sound good on them, rather than good in general.

The TV and weather bands are only available on the US model.

I've never used the a unit with VPT but concensus is generally that it's useless. It's not the type of feature I'd use even if I had it. I don't use even a third of the features built into the 700, actually. I've never bothered learning how to do the more advanced navigation type stuff, for instance, because I have no use for it. I simply know what's on my discs, and how to get there with basic controls, which is good enough for me.

Addendum: the NH700 also has the 6-band EQ. Comparing it with the NHF800, all it lacks is the tuner remote and VPT engine.

Heh, I think not. What's available there is actually my *full* collection of "prepared" recordings, i.e. the stuff I've already edited et al. There are other recordings.. Such as another of Joshua Stanton, done with the binaurals but ruined by the fact that he used a micro guitar amp for his sitar, which simply sounded like crud.. And a couple of poetry readings, one of which takes almost 2 full CDDA discs, the other being about 35 mins long, but not containing much that anyone else but those who were present would have any interest in. I tend to avoid recording live shows, actually. Unless it can be done 100% acoustically, i.e. with no amplification, or with balanced individual instrument amps in an acostically-controlled environment, I don't see much point in the effort. The results simply aren't what I want them to be. Basically, I'm not so interested in doing bootleg-type recordings. The Backpedal show [also featuring two other bands, one from Edmonton] was my one real foray into seeing how well it could do, given the horrible space it was recorded in and the fairly minimal PA that was used. I'm more interested in recording things that sound natural, and do so, well.. naturally. Ideally, what I'd like to get are some recordings of folk musicians sitting around a campfire or something. All acoustic instruments [guitar, mandolin, standup bass, fiddle, hand-drums, voices..] with some minor arranging as to placement around the mics for balance.. much more like most of the jazz recordings done before the late 1960s or so. I'd also love to do some jazz recordings. There is a top-notch jazz department at the conservatory here [The Queen Elizabeth II School of Music, to be precise, part of the Brandon University campus] and I've known -many- of the players who've gone through their programmes over the years. Unfortunately, I've gotten old and out of touch, as well as cranky and unmotivated. At any rate, those are pretty much the extent of the recordings I'd bother to let anyone else hear.

This is what additional mixer channels are for. If you want to mix two stereo mics, that's 4 channels. Ergo, you'll need a mixer with at least 4 channels. In the example I gave, each mono plug from the y adapter is a side of the mic, and each one would then have its own plug. One mixer channel per mic [side], then.

Note that you have to use the user id and password I supplied above [right beside the URI, in fact]. If you're having problems seeing my end at all, I'm not sure what to suggest. Other than the simple password security, there's nothing odd about how I run my webserver. Unless your end isn't catching the DynDNS address correctly.

By default, the HiMD recorder will use HiMD mode regardless of which type of disc you put in. It's more important to select a mode that is appropriate to the length of the recording you'll be making - PCM for shorter than 95 minutes with a 1GB disc, or 28 minutes with an MD80; HiSP for up to 7 hours 55 mins on a 1GB disc, 2 hours 23 mins on an MD80. Normal MDs require slightly less power to write to. Make sure your battery is freshly charged before you go to make the recording. With a mic like the 907 I would recommend actually sticking it in the piano if at all possible. Close-mic'ing with a stereo mic as high above the strings as the lid allows, nearer to the centre of the sounding board, will probably get you the best results. I wouldn't try mic'ing from 10' away with a 907. This mic position can be easily achieved using a standard boom-type microphone stand with a rubber clip that the 907 fits into [shure and Peavey rubber mic clips both fit the 907 just fine]. Put the stand on the open side of the piano [assuming it's not an upright] with the boom extending in towards the lid, with the mic pointing down at the strings just past the hammers. I'm assuming you'll be allowed to do this, of course. Use mic sensitivity set to LOW. Set levels manually by getting the pianist to bang around for a minute. Set levels so the loudest peaks meet the dash [-12dBfs] on the record metres. This gives some headroom. With the 907's sensitivity the quietest sounds the piano can make will probably still be drowned out by ambient room noise, even when close micing, so don't worry about losing the quiet parts too much.

Notes on the radio drama: The story was written by playwright Dale Lakevold. Music was ripped from various sources, most of which are Canadian. The basic story is that of the flagmaker, an artist in the making, and how he became the flagmaker. Most of it revolves around various points of interest in Canadian history as well as bits about where he goes on his travels. The subwoofer track is messed up because of the system that was used for monitoring. All the bass from music tracks had to be filtered off to a separate mono file to be played on the sub track, to enable proper panning of stereo channels around the quad soundscape without the bass phasing in and out completely. Chances are, it will have to be turned up to be listened to properly. Blame the logitech z680s. To those of you are aren't Canadian [almost all of you, I'm sure] most of the historical and geographical references will be meaningless. Still, you might find the piece [56 minutes total] interesting at least.

I have thrown all kinds of VBR mp3s at SS, as well as WAV files that were at sampling rates up to 96kHz at 24-bits. It handles most of them correctly [most notably, all WAV formats I've tried importing do so successfully with proper/adequate sample-rate conversion and at the very least bit-depth reduction that works, if not in fact full use of 24 or 20-bit data at the encoder]. The exceptions for me seem to be mp3s from certain encoders, which come through only as silence - regardless of whether they are CBR, VBR, &c. I wish Sony would just start using default Directshow/Directsound filters instead of their own codecs for formats other than atrac/3/plus. This would make SS capable of transcoding any format you have a directshow filter installed for, from mp3 to aac to ogg to MPC to AC3.. Most of which carry no DRM of their own. If they're going to let people transcode their mp3s, they should also let them transcode anything else they have the codec for. Caveat: since there is no real standard for how codecs pass metadata [title tags &c.] this would make it even harder to get titles and such passed to SS. Not impossible, though.

Since I hardly use my web server for anything any more, here are some of my recordings. http://dexotaku.ath.cx/sound-art / uid: mdcf / pw: mdcf Folkfest and Soundscape: These recordings are some that I've done around the area where I live, as well as tracks done at last year's local Folk Festival. All of these were made with my NH700 in either PCM or HiSP mode, using SP-TFB2 in-ear binaural microphones. Gentle normalisation has been applied to some of the tracks, but in all cases I tried to do this while also preserving the original dynamic range. -12dB/oct cut applied at 145Hz to most tracks. [This filter lets things sound more 'normal' over speakers, because otherwise the bass is next to deafening - though on headphones things sound normal. Weird, aye?] The Folkfest tracks are heavily normalised in parts. My apologies for this, as the FOH PA was rather inconsistent in how loud some instruments were, as well as when people were speaking. Advice: try the long trainpass track. At the beginning there is a stretch of ambience in the railyard - turn up your system until the crickets and the whine of a hydraulic track switch seem "natural" to your ears. Then listen to how loud the train is when it comes [from the right]. Please don't blame me if you damage your playback equipment while playing this. It has been peak-normalised, but otherwise the original dynamic range represents as close to exactly what I heard as can be recorded with this equipment. [i.e. the dynamic range is probably about 100-500X that of most music recordings]. Backpedal: a full FLAC image [with a cuesheet] and to the MP3s of that FLAC. This was a show recorded at a small local pub, a band playing through a PA. Also included are lots of tracks of before and after the show, wandering in the streets, &c. Parental advisory: Some of the recorded conversations in this lot involve mature themes such as recreational drug use. If you're opposed to or offended by this sort of thing, please don't listen to it. The Backpedal show [see http://www.backpedal.ca] was also recorded with my NH700 and the SP-TFB2 microphones; manual levels during the show, AGC on for the street and ambient passages. All recorded in PCM, and again, a -12dB/oct cut applied at 145Hz for music parts. The Bitter End is among the better music tracks in this one. j.stanton: Joshua Stanton [no website] is a sitar player from Winnipeg, Manitoba. His material is mainly hindustani classical or pieces he's written himself which could be viewed as also in that genre. This particular piece [29 minutes long] was originally written for [but not used in] the Guy Madden film, Saddest Music in the World. The recording was almost totally impromptu, is in mono, and was taken from the tape output of a Peavey XR-680 amp/mixer. I didn't expect much in terms of quality when I made this, since I had no control over balance between the sitar pickup, its mic, and the mic on the hand drum being used by Joshua's musical cohort for the night. I was rather surprised by how good it ended up. 8X stacked -24dB 1/3oct notch filter was applied at 60Hz as well as 120 and 240Hz to kill the louder harmonics of hum caused by the amp. I have some more to add to this when I'm done editing. Do not expect high-speed downloads. This is running on a residential DSL line [384kbps upstream] which is also being used by one filesharing service. Your patience is appreciated. Enjoy! edit: Added more tracks that I recorded in December to the Soundscapes folder. These include sounds from my father and I shooting pool, as well as two of the elevators [including the silly one that talks] at the Brandon Regional Health Centre. Also: feel free to use the non-music recordings in edited pieces if you like. Just credit me as the source of the sounds. As far as the music recordings go, the folkfest ones should not be widely distributed, though the Backpedal recordings as well as that of Joshua Stanton may be distributed as you wish - once again, give the artists credit - i.e. leave the tags intact, if you please. edit2: Lastly, another addition. I threw the folder of various versions of the 5.1-surround 'radio drama' I did for an art gallery installation in 2003 on there as well. The only "accurate" version is the AC3 [Dolby Digital] one which is in full 5.1-surround and can be played standalone on a computer with software like VideoLAN Client or Foobar2000. If you have a surround-sound system with your 'puter, use these ones if you're interested. Users with DVD recorders can make a standard DVD with a blank video track and this for audio that should work on any player compatible with whichever format blanks you use. The rest are stereo, dolby-stereo [for Prologic decoders, though I must admit that the encoding is often flaky], and mono [was mixed for playback on CBC radio 1]. These are all standard mp3s and will play on normal equipment, of course. Note that the surround project was not recorded on my HiMD [of course]. Most of the sound effects were recorded using a MZ-R70 standard minidisc recorder and an MS-907 microphone. [other sound effects came from the BBC SFX library, ripped from CDs kept at the local university, yay!]

The thing about live recordings like this is that the recording is -not- of the band; the recording is of the -room-. Stereo mic'ing a band playing through a PA is actually a next-to useless exercise in terms of trying to capture the band in stereo. For one thing, most live PA systems [especially smaller ones such as those permanently installed in clubs] run in mono only. The only way to get a band in true stereo is to either have them all play acoustically, or to set the instruments up around the mic and run without a central PA, only with instrument amps [whose physical locations make up the stereo image]. Otherwise, with this recording as with almost all others in done indoors and of PAs, all the stereo effect of the mic [is/are] good for is the sense of "being there," which, even with M/S and coincident-Y mic arrangements, still sounds better on headphones than over speakers [iMHO]. Judging the quality of the recording in these cases is very difficult, since what's being recorded is actually kind of shitty, really. What you end up with is a very accurate recording of occasionally unlistenable sound. Being able to monitor and choose mic placement based on actual listening inside the space [test recordings or isolation monitoring] helps a lot. Sound check is perfect for this, especially if the FOH engineer and the band know you're recording and are co-operative in giving you a minute or two to set up. Of course, it's next to impossible to do this when stealth recording. I no longer consider concert bootlegs to be what I'd consider "listening music." The only purpose they serve to me is to get an impression of what a band is like live, what kind of energy they have along with the crowd's. The immersiveness of most live recordings [being made from the crowd] can provide a real sense of "being there" that transcends the technical problems that almost always occur - shifts in mic position, arseholes yelling in the recordist's ear, &c. While I haven't actually listened to the recordings yet, that is the standpoint I'd be looking at them from. In these cases, I actually rarely consider the fidelity to be of crucial importance - it's very difficult to get good-fidelity let alone high-fidelity recordings of live shows without having some kind of mix straight from the sound board to mix with ambience. Basically, my rule of thumb is this: take it for what it is. Which is why I'm often surprised by how good some of the bootlegs that people make with MDs and a $100 microphone are.

If you get a Y adapter, you can split the output of the mic into two mono channels. This is assuming it's a self-powered or unpowered mic, since the mixer probably doesn't supply plug-in power. What you need is this: 3.5 stereo jack to dual 3.5mm mono plugs

Kudos to you!