greenmachine

When I was suggesting a level setting, I was assuming a similar scenario. i.e. recording the same band in the same location from the same distance via the mic-in at low sensitivity setting (you want to use this setting most of the time, particularly for anything that is louder than an average conversation). As I understand it, the AGC sets the level approx. comparable to a manual level setting of 20 and automatically corrects downwards where necessary to avoid clipping, at the cost of destroying the natural dynamics (sharp attacks will be softened). Since the AGC seemed to be active only every now and then, I thought a relatively small correction of the level setting downwards from 20 might be sufficient (one step equals approx. 2dB). The Sony mic you're currently using is a good beginner mic . It has the advantage of a relatively low sensitivity (at the cost of weak bass response and below average S/N), which is useful when recording loud sounds via the mic-in. With a higher sensitive mic, the preamp may overload no matter of the level setting and you'd be better off recording via a battery box through line-in (a more complex setup, can cause additional annoyances like and automatic track marking). Single point mics also have the advantage of being relatively easy to set up. You may want to stick with this mic for a while as you're learning the basics of live recording and upgrade anytime you feel confident.

Not 100% sure about the circuits used in this particular mic, but I don't think the voltages of internal and external battery would add up, you'd most likely still have 9V.

9V is the minimum for this non-prepolarized condenser mic. It will most likely perform better with higher voltages (up to 52 Volts) via phantom power..

I think the problems you hear in the recordings are dynamic compression artifacts caused by the use of the automatic gain control (AGC). The "for loud music" (instead of the "standard") setting may perform a bit better for music, but for the most accurate results you may want to use manual levels. The "best compromise" setting depends to a great part on the sensitivity of the microphone, the mic position, the loudness of the signal to be recorded and the room acoustics (among other factors). You'll need to experiment a bit to learn to estimate a good setting with your current equipment. For similar gigs like the ones shown in the samples, start with a setting of approx. 15 or 16 (of 30) and work your way upwards or downwards, depending on if the recording is becoming too quiet or there is clipping. It's usually better to leave some headroom (peak level below maximum possible level) than to risk clipping. It's usually better to set the level before the session and leave it there instead of re-adjusting during a recording.

3 and 4 are used as opening switches - if you don't need them, simply don't use them. You could as well connect 2+3 and 4+5, it would make no difference for your purposes. The cable in the picture is usually used for headphones, not sure if suited for recording - the individual cables are lacquered instead of wrapped in plastic. If you want to use it, you need to scrape the lacquer off at the ends, otherwise you will have a hard time trying to solder them to the board. Not sure about the colors, use an ohmmeter. As you have figured out correctly, the outer ring of the plug is left, followed by right and ground.

Just because the unit is marketed as USB 2.0 compatible, it doesn't mean it uses the maximum possible USB 2.0 speed. In fact, it is far from it - around USB 1.1 speed or just slightly above (USB 1.1 = 1/40 the speed of USB 2.0). The 100x transfer speed figure is only true for highly compressed low quality audio (48kbps).

I have no RH1, but it works on each of my three other recorders (NH700, R909, R700). I don't see a reason why the RH1 should be different. Give it a try.

There should be some sort of description coming with it - if not, connect a cable and use an ohmmeter to figure out what belongs where. On the cable, the left channel is usually coated white, the right channel in red, while the other one is ground.

A guess: The battery was still partially charged before having started another charging process. Older MD models don't have a "smart battery full" detection, but stop charging after a fixed time interval (timed charging). It is easy to overcharge if you're not careful, which is what I suspect happened here. Ni-MH batteries should be in a close to empty state before attempting any charging process, particularly if there's no smart detection as with later Sony models.

The disk door, as with all MD recorders, should be automatically locked during any recording or editing processes. The eject button will have no effect. During playback there's no lock though.

I don't get the fuss about mono compatibility of stereo mics. If you know you want to record in mono, why not just point one of both channels towards the source and discard the other channel in post processing (instead of mixing them, which can lead to phase problems indeed)?

Virtually all stereo jacks have essentially three contacts - left, right and ground. Whether these are to divided into 3 or more pins doesn't really matter, you just need to know the layout (where each pin leads to). The one i used in the sample pictures has 4 pins, in this case the left channel was divided into two pins - for a better fit onto a circuit board, i guess. (please excuse my poor german-english, not good at explaining technical matters)

Dunno, the samples work at my end (using firefox with quicktime for in-browser mp3 playback), not sure if they're donloadable, maybe try a different browser. I am quite satisfied with the hrtf placement (with the mics close, but not in your ears, directed forwards) and used that most of the time. If we had places with an open tapers section, i might be tempted to experiment with a jecklin disk or dummy head or cardioids in DIN config. You may find this site interesting, describing microphone techniques. By the way, i guess you know about the tapers section? Lots of info regarding mics, techniques, jecklin disks there.

I've been comparing various kinds of mic placement (for omnidirectional mics like yours) a while ago including two differently sized jecklin disks - the thread is here. I've done the recordings in a rather small living room, recording music from loudspeakers.

...at the cost of compression ratio/filesize. Any further conversion to any lossy codec/bitrate will be another lossy step, even if re-recorded to SP mode. Oh, if they only...

In the library view of sonicsatge, right-click the track(s) and choose "save as wav".

SP and Hi-SP are different codecs at different bitrates (292/256kbps). A conversion from SP to Hi-SP will be another lossy step. Alternatively you can decompress them to wav and losslessly compress these, but compression ratio will be worse (flilesize will be larger). If they only could be uploaded and further used 'as is' (in SP)...

Search the forum for "TOC cloning".

... which would be still less than USB 1.1 speed (12Mb/s = 1.5 MB/s). How much higher is the read speed?

I wonder, how do you people usually proceed with unwanted track marks caused by line-in recording? On the unit editing, despite the increased reliability of later SS versions seems still risky, while combining via SS (after uploading) seems so incredibly resource-heavy.

If you think about it, it seems quite logical that on standard MDs the read intervals need to be longer or at a higher speed or both: There is less information to be read/written, but spread onto the same physical space, or in other words, the read/write mechanism must work its way through a whole disc in less time.

Non-polarized ones will work at least as well, but are usually a bit larger. If you don't care for the maximum possible miniaturization, use them!

I don't think the very gentle spinning of the disc causes enough vibrations so that it can be easily felt, but the motor noise can certainly be heard in a quiet environment. It would be interesting to see if the RH1 behaves the same way, or if they maybe increased the buffer size.

Geri Allen - Eyes... in the back of your head

The buffer in (Hi-)MD units (and portable CD players) is in fact a small RAM (Random Access Memory), like the one in your computer. Every couple of seconds or minutes it is refilled with information from the disc. The audio that is sent to the headphone output is not directly read from the disc, but from the RAM. As long as there's new information in the RAM, there's no skipping, no matter how hard you shake the unit since there are no moving/mechanical parts in a RAM. Initially (when starting to play back a disc), the RAM is filled completely with data from the disc, which takes a couple of seconds. After that, the disc mechanism is idle until the RAM reached a certain degree of emptiness*. It wouldn't be very smart to refill the RAM only when completely emptied since the unit is susceptible to shock in the disc reading process, which would cause immediate skipping/dropouts. If the buffer is refilled before completely empty, there is some buffer left in case the disc can't be read immediately (shock). If the unit is under constant serious shock, the RAM can't be rewritten, which will bring playback to a halt eventually. If the shock occurs periodically, there's usually time for the buffer to be rewritten before it runs empty. The RAM has a certain size. The lower the bitrate of the audio (less information per time period), the less often the RAM needs to be refilled. The lower the bitrate, the more efficient the anti-skip mechanism will work for several reasons: - The speed of the disc drive is limited, it takes longer to read a 30 second PCM file (1411kbps**) than it would take to read a 30 second Hi-LP file (64kbps). - It is sufficient to refill the RAM at a later stage (higher degree of emptiness) with a low bitrate file. A certain buffer time will take less space on the fixed-size RAM. Information needs to read less often from the disc. On my NH700 and a 1GB disc, the read/refill process takes about 8 seconds. In PCM mode, the disc mechanism is idle for about 30 seconds before the next refill process takes place. In Hi-SP mode slightly more than 3 minutes, in Hi-LP mode just over 14 minutes. I suspect the RAM to be 8MB in size since a 14 minute Hi-LP (64kbps) file would be roughly 7 MB large and there must be still some buffer left (as explained above). *empty = no new/relevant information to read **kbps = kilobits per second