dex Otaku

daniel: there is no way to upload directly with legacy MD / MDLP discs using portables; HiMDRenderer is usable only with recordings already in one's SonicStage library [i.e. converted by SS or uploaded from HiMD]. Please take a look at the thread I linked to.

There is no way to do this directly. See here: http://forums.minidisc.org/index.php?showtopic=7070

HiMD supports only the following bitrates: HiLP 48kbps HiLP 64kbps LP3 105kbps LP2 132kbps HiSP 256kbps a3+ 352kbps [added with SS 3.3] LPCM 1.4Mbps No other bitrates of atrac3 or atrac3plus are supported by HiMD.

Knowing pretty well what flat sounds like, I usually set my EQ to meet approximately that and then compensate a bit for my hyperacusis by cutting the 2nd highest band slightly more than the rest. A lot of the music I listen to I'm already well aware of the sonic attributes of [with some after more than 20 years of listening], so I know reasonably well what to compensate for. I think posting my EQ settings would be pretty useless though, because they're completely different for each set of 'phones and even with my 'general' setting for each set I still compensate for individual albums according to my personal preferences and the need to avoid pain caused by hyperacusis. I don't see anything wrong with EQs either, myself. I personally never use them for additive processing, having come from the school of "passive is better" and actually having the ability to hear clipping distortion caused by EQing too much in the positive in any given band.

Heh, "old mood"? At high levels into the decoder stage, sure, C-type NR shouldn't have too much artifacting in the sense of de-emphasis, but then - at the highest levels, it also does little to no expansion, either. One of the premises of most noise reduction schema [dbx compansion being an obvious exception] is that (post-emphasis playback-processing) high-level signals approach an expansion ratio of 1:1 at their upper threshold. The end result of putting AGC'd material through this kind of playback processing should in theory be that in quiet parts the high end is quite audibly lost [up to 20dB with Dolby C], and in really loud parts, no expansion takes place in the range where you actually want it to [the peaks]. Mind you, that's theory based on my understanding of things [mostly based on things like this: the Dolby labs white paper on B, C, and S-type NR]. Both my theory and my understanding are potentially flawed. The method itself is kind of cool, actually, and we all know that practise differs greatly from theory. I'd actually like to hear a before and after of the real-world results. [No, I don't currently have a cassette deck with better than B on it.]

SS can convert any track uploaded to SS from yur HiMD that was either a microphone or line-in recording. HiMDRenderer can do the same job, as well as converting any track with valid rights in your SS library. So yes, you can use it to convert tracks you purchased from Connect.

That's an interesting method suggestion, killroy. It has one unfortunate side-effect, though: Dolby B, C, and S all use bandpass compansion, C far more obviously than either of the others [above above 10kHz], so the end results would be a completely flattened high-end at the very least.

AGC = compression/limiting. Compression means: when levels go above a certain point [the threshold], they will be lowered by a certain amount [compression ratio]. In its simplest form, straight 2:1 compression say, above -12dBfs [that's the middle mark on your recorder's level meter], means that at 1dB above -12 it's compressed 2:1 to give, well, 1dB above .. at 2dB above, it's compressed 2:1 to give 1dB above .. at 10dB above, it's compressed 2:1 to give 5dB above .. and so on. Limiting is infinity:1 compression, meaning that anything above the threshold is reduced to the threshold itself. AGC is usually a combination of the two, i.e. compression above a certain threshold and then limiting just below the maximum level so that hopefully clipping distortion never occurs. Another possibility for AGC is dynamic compression, where the higher the levels go above the threshold, the higher the compression ratio gets, until it reaches limiting. Expansion is the exact opposite: when levels go above the threshold, they are -increased- according to an expansion ratio. Technically, there is no difference in the process involved between expansion and compression except that instead of being 2:1, it's then 1:2 coming out. I am not adept at explaining these things, though. If you're interested in learning, a much better explanation can be found at wikipedia.

It is theoretically possible to take an unprocessed copy of the recording and apply expansion to it to compensate for the AGC. It won't work 100%, and to really do it properly you have to know the threshold, attack, release, and ratio that the AGC works at to attempt to apply its reverse. These are not listed in any of the manuals. The only way to really find out what the values are is to repeatedly run test signals through the mic input of the recorder to try and figure them out, something which I honestly wouldn't want to invest the time in doing myself. You could fudge it by putting expansion [which is the opposite of compression] on the recording and just fiddling with the settings until you get something that sounds less-bad than your original. Good luck..

AAL appears to work the exact opposite of what you'd expect, making it almost completely useless. The useful way would be for it to have a lossless copy along with a compressed copy at your most commonly used bitrate in one file, so when you go to download it to your player it doesn't have to transcode... and if you want to download a different bitrate, it would use the lossless copy as the source for trancoding. The lossless copy gets used for playback in SS at all times. It does -almost- all of that, except for one thing: if you want to download at a bitrate other than the one you selected as your "default" compression, it transcodes the compressed copy rather than the lossless copy. What this means is that the only use for the lossless copy is for listening from SS; it doesn't get used for anything else. If you store things in your library using WAV/uncompressed, it transcodes to whatever bitrate you select from the uncompressed source, meaning it's always using the best possible quality the codecs can produce.

While the headphone amps of the various models vary [being either analogue, digital, or high def digital], all of the models feature a 6-band EQ. That said, you would be far better off investing in a good pair of headphones that suit your needs than in trying to compare the bass output of the various models [or, for that matter, any portable].

True, though [unless the source is PCM] you're incurring at least one generation loss in the process through transcoding, and have to reconstruct track info on way or another.

DRM is made specifically to prevent what you're trying to do.

All current HiMD models that I'm aware of have a 6-band EQ built in, rather than mega bass. See the equipment browser.

I really need to wake up. I read the post but not the title - you're transferring mp3's, so the bitrate from SS shouldn't be an issue as long as they're not audiobooks or something. That's assuming you're transferring them -as- mp3's, of course. I still don't understand what you mean though. What exactly does something that is "buffered" sound like?

Both coax and optical [toslink] carry the same signal. Coax S/PDIF is the consumer version of the balanced ASE/EBU digital interface; toslink is the same thing only done with fiber rather than coax. The data in both instances should be identical, and is a subset of the AES/EBU standard. Look up SP/DIF on wikipedia for more complete info. Given the short cable runs that most people have with home equipment, the difference between them should be nil. I have had friends claim that optical is vastly superior, so much so that they could actually hear the difference [with ordinary consumer stereo equipment] which is highly unlikely. If you are able to discern any difference between the two from the same source, it would usually [i would assert almost always, actually] indicate faulty equipment, not the inherent superiority of one over the other. Coax technically should be less prone to timing errors, as I understand it, because of the nature of optical cable. From what I recall, it can build interference patterns like standing waves when the fiber is curved too much or at specific angles. The main advantage usually touted with optical cabling is isolation from electromagnetic interference. Really though, with most people's cabling being no more than a metre in length, there should be no practical difference between coax and optical unless you're trying to bend that fiber around loops and sharp corners.

Check what bitrate [encoding mode] you're using. If it sounds like that, you've probably got it set to encode in HiLP mode, which indeed sounds pretty similar to internet radio as it's a very low bitrate. On the other hand, your description [now that I've reread it] .. "hacked and buffered" .. a teensy bit vague. Do you mean it's playing back choppy, with gaps in it?

This is really interesting. Thank you for the boot to the head, DerTapir! I don't know why I never checked my NH700 service manual for this. The NH700 shares the same ADC as the NH1. As ATRAC itself was designed for encoding up to 20-bit sources, it's fairly likely that the internal encoder of these portables is capable of encoding ATRAC/3/plus directly from the 20-bit output of the ADC. If this is true, it's possible that the dynamic range of ATRAC/3/plus recordings could in theory be higher than that of PCM recordings. I have no way to confirm this. Perhaps if the encoder supports 20-bit data, someone could try sending a 20-bit stream via SP/DIF to their recorder via the optical in and see what happens in various recording modes. It would appear that our recorders have an actual linear converter in them, finally smashing any theories I had about cheap 1-bit converters. Thanks again, DerTapir. Also, the RH10 has the same ADC. Without checking all of the service manuals, I'm assuming the same chip is used in all current models.

First you upload the tracks to your computer with SonicStage. This makes a copy in Sony's proprietary [i.e. next to useless] format. Then you export them to .wav and you can do with them as you please. Check the uploading FAQ, more detailed answers are there. Yes, it works as removable storage, but it's very slow.

Your English is definitely better than my Italian. No, not really. Evidently these sales people don't know what they're talking about. HiMD [2nd-generation] can record in three modes: HiLP [64kbps], HiSP [256kbps], and PCM [exactly the same format as CD audio, uncompressed]. You can upload recorded tracks using Sony's software, SonicStage. SonicStage itself isn't the most ideal software on earth, but considering the cost difference between HiMD and its nearest competitors, a few minutes extra work is definitely worth it.

Do some research on the microtrack before you buy. Going by most current reviews, the MT24/96 is basically a product still in beta; the software [on the unit itself] is both unstable and unreliable as described by most. The quality of its mic preamp and a/d conversion are also suspect. Though I have seen no actual technical comparisons, I have read that HiMD at 16/44.1 still outperforms the mt24/96. You don't have to take my word on it, though. Google "microtrack 24/96 review". Some interesting ones, both positive and negative in content: http://www.sonicstudios.com/mt2496rv.htm http://leblog.exuberance.com/2005/10/review_of_the_m.html http://www.dvinfo.net/conf/showthread.php?t=51631 which also refers to this one which appears to be among the best: http://taperssection.com/index.php?topic=50364.0

Another simple thing to try, before reinstalling, is to make sure you plug in the NH700 -before- you open SS. I've had SS take its time detecting units [and rarely not see them at all] when I plugged them in after SS was already loaded. That aside, I'd back sebastianbf - remove SS and its related components, and install the newest version [currently 3.3]. Please take a look at this thread on reinstalling before just going ahead.

HiMD suits your needs as you've listed them. You can compare some models at the minidisc.org equipment browser. There are also reviews and such here on the fora. It also may be useful to skim through the essential HiMD info/FAQs forum. Some of what's there won't make sense unless you already have a unit to experiment with, but others, the upload FAQ in particular, can give an idea of how HiMD works. What's your price range? Cheers.

To be frank, I didn't find the comparison very useful. When doing a comparison like this, the lowest common denominator should be the one that is most directly compared, i.e. all equipment in the highest sampling rates and bit depths that -all- of them support, i.e. 44.1kHz, 16-bit. You should also use exactly the same mic with exactly the same preamp set to exactly the same gain for all tests with all equipment. The movie itself just confuses this by putting things in a completely nonsensical order that basically makes direct comparison impossible. I admire the effort of the tester but find his methodology lacking. The results are basically useless.

Also: turn the volume up to 28 or 29/30, and disable the built-in EQ completely [sound: normal].