greenmachine

I know, Im using these as well, but what is a binaural MONO mic? Not if you recorded it with a non-Hi-MD unit, what you obviously did. But if you have a MD deck with digital out (optical/coaxial) and a soundcard with digital in, you could transfer it digitally in realtime. By the way, we must have a largely different perception of loud, since I got about 4 bars at 20/30 through line-in with not too sensitive microphones at a loud concert...

Not at all, at the moment you only seem to be able only to choose between the 5 latest albums and the 1-5 first, nothing in between. Although I've found a way to work around, there's still scope for improvement.

Maybe a level discussion should be added to the important topics, since this question seems to be brought up pretty often... What is a binaural mono mic by the way?

Sony's level meters usually have 9 bars, where the distribution would be approximately as follows: 1st: -40 dB (or less, on some units it's always on) 2nd: -30 dB 3rd: -20 dB 4th: -12 dB 5th: -6 dB (first mark/dot, try to peak at this one if you don't know the absolute peak level of the source) 6th: -4 dB 7th: -2 dB 8th: 0 dB (peak at this one if you know the absolute peak level of the source) 9th: over (second mark/dot, avoid this one at any price!) If it only hit the 3rd bar, you could have gone up to 20 dB higher. A too low level usually means increased noise (after normalizing). Additional noise could be introduced by analog transfer to the computer. Are you transferring digitally? The best method to clean noise from the recording would be a sustraction [(signal+noise)-noise=signal], but you need a signal-free passage from your recording, which contains only the noise of your equipment. Adobe Audition for example can perform this noise substraction and has some decent delay effects.

Alright, I've tried it with a different source, a CD player with optical out - still no signal... Maybe they've sold me crap. I need to test it with a different device, maybe a DAT deck with coaxial in...

I own a MD deck with optical output (MDS-JE500) and a soundcard with coaxial input (Creative Soundblaster Audigy 2), so I decided to buy an optical to coaxial converter, connected cables as suggested, powered the converter, selected s/pdif as recording input, set a proper volume in the mixer, started the MD player, opened a recording program, pressed the rec button, and ... nothing happened. Am I missing something here or doing sth. basically wrong? I thought it would work the same way as the analog method ?!? How can I test the converter, as I am not sure if it's working?

MD recorders are pretty delicate magneto-optical devices, they should better be left motionless while recording, they are likely to produce silent spots or crackling sounds when not treated very carefully. Maybe a hard-disk- or flash-based recorder would serve your purpose better for rough street recording.

It's exactly the same thing, Hi-SP is Hi-SP and uses Atrac3plus, whatever disc you're using, you'll just have significantly more recording time with a Hi-MD. Recording in old 292 kbit/s SP Atrac is no longer supported by Hi-MD units, just playback.

Speakers diffuse and 'color' the sound by involving surrounding room accoustics (which can be positive for very 'dry', direct sounds), whereas headphones don't.

It's not just about bitrate, it's also very important how you throw 'unnecessary' bits away, frequencies you are supposed not to hear, this is why all these different lossy audio codecs like mp3, ogg vorbis, mpc, aac, wma, atrac, ac3, real audio and so on, exist. I believe Sony claims the new Atrac3plus to be even superior to their older Atrac Type whatever, despite it's slightly lower bitrate, although it has yet to be proven.

--alt-preset standard is a x-times proven Lame setting for 'transparent' lossy compression, in most cases undistinguishable from the uncompressed source. It is variable bitrate (VBR), which means that it adjusts the necessary bitrate for different sources automatically, easy to encode tracks give a lower, hard to encode tracks a higher bitrate. The average resulting bitrate would be approximately 190 kbit/s, about 1.5 times as large as your actual 128 kbit/s samples. You can read about recommended Lame settings here and the corresponding discussion here . 128 kbit/s CBR can be good enough in some cases, but use at least mid-/side joint stereo mode mode then, it means to losslessly couple the channels if possible and this way improves the sound. For a more detailed discussion, read here or here or here or simply search for "joint stereo" at the excellent audio compression discussion forums at http://www.hydrogenaudio.org/ . Hope this helps.

That's a pretty unusual decision, I'm looking forward to the results.

A excerpt from the site: Hmm...

Here's the "professional" solution, although I wouldn't spend 35$+shipping for two resistors, two plugs and a cable...

I believe these ones not to be significantly worse, if at all, propably they even use the same Panasonis capsules with the same modification, although the core sound's are declared to be closely matched, but in the end it's your decision...

These seem to be a very good handmade version, seems to be in the same league as some much more expensive models. I'd buy them if they didn't already have some, although i can't imagine the pain when someone is trying to clip them to his ears ... Strange, it was a huge difference for me, but as I've already said, it has an additional effect to put something between them, try to wear them near your ears for example and listen to the results through headphones afterwards.

What's wrong with me? Since my first successfuly recorded live show I can't listen to all these flat-sounding, compressed-to-death studio recordings anymore, at least with headphones. I'm feeling fooled. It's really awful. It's like an awakening... Anybody out there who feels the same?

Yes, just samples, i know, but how can you present someone your recording efforts if the quality is significantly degraded by a poorly set encoder?

@Wedge: I think it could sound awesome if you could encode it as higher quality mp3, maybe --alt-preset standard with Lame, or if you have to use such a low bitrate, use joint stereo mode. You can find a very easy to use frontend using the high quality Lame encoder here. Just drop your wav files at it.

I'd say they're both pretty bad and close to unenjoyable, although I generally perceive the distortion caused by an overloaded mic not as aggressive as a clipping preamp, but listen for yourself... It's difficult to get a distortion-free recording in such environments (small room + loud PA) without a battery box, but I've propably exposed myself and the equipment to even higher SPL's than you did and for that reason didn't succeed with an attenuator... Ideally as high as possible without hitting the max, but better somewhat below than above. But if you'll get too low, you'll experience increased noise after 'boosting' the level afterwards (normalizing), at least with direct microphone to line-in recording - it works for using the built-in mic preamp with a fixed/selectable gain (low/high sensitivity) though. In my experience it made no sense to go below approx. level 10 when using the built-in mic preamp though, the distortion couldn't be reduced any further below that level setting. By getting higher than that level, you just limit the dynamic range by raising both the input signal together with the noise floor. Of course this point depends on your mic specifications. When using the built-in mic preamp, you can try to find that 'sweet spot' by exposing the mic to a high continuous SPL so that there's just a slight distortion with a rather low recording level setting (anything below 9 or so should work), then raise the level until it starts to distort even more. You can use the the highest level setting before the increased distortion starts to happen for a convenient universal recording level and amplify it afterwards without significant side-effects. Basically it means to use the limited gear effectively. I would use higher levels only if I knew for sure the source has a strongly limited dynamic range or if I had no possibility to normalize it afterwards or if i couldn't upload the material digitally. The qualitywise best solution would be of course a good mic preamp with adjustable gain plugged into the line in, but I couldn't find an affordable yet.

It shouldn't do any damage but draining some battery power, i believe earlier md units were spinning the disc while in pause mode, nowadays they don't, at least the newer sony's.

There's one more problem with the attenuator: You better have to use it exclusively in max. position, or you'll risk to reduce the already low voltage supplied from the recorder to the mics, depending on how the potis are connected.

I'm going to very loud shows too, and the attenuator reduced the distortion to some degree by relieving the preamp from loud input, but then the microphones themself were still overloaded and I still had bad distortion, so I had to build a battery box and from that point it worked beautifully. The attenuator may be suited for a little bit louder than usual sounds, but not for noise earthquakes... I don't think standing further back from the stage in a rather small to medium room with lots of bass helps a lot, if at all...

Hehe, it's the loud high-frequencies which are the most dangerous for your hearing abilities, not really the bass...

It could, but there are microphones which can handle much higher SPL's with virtually no distortion than your ears are able to. Hehe, making a record is easy, making a good one can be a little bit more tricky, but it's well worth it in most cases...