dex Otaku

It's a rounding error. There are two standards for showing kB/MB/GB and the like. The 'correct' one of computer capacity terms uses powers of 2, so 1B = 1024 bytes. The other, less honest standard uses 1000 bytes as 1kB. As a result 1MB does not = 1MB. [This is much the same as why hard disc manufacturer's reported hd sizes are usually totally incorrect - the actual capacity is always lower than what the mfg specifies simply because they're rounding from powers of 10 rather than 2.] The short version of all this is that a blank 1GB HiMD has less than an actual 1GB free on it; it's pretty close to 1,000,000,000 bytes, whereas 1GB = 1,073,741,824 bytes. There are other thins to take into account as well, such as the DRM table, track info database, and the like.

Give the choice, I usually go from the PA. I don't particularly like stealth recordings of performances amplified via PA, to be honest. This is keeping in mind that I usually have access to the PA and usually know the person running it [if it's not me]. In the past I have split the Mackie PA that generally gets shuttled around here by using its bussing system. Here's how I generally do this: * assign all the mic and instrument channels to bus 1/2 and 3/4 * assign bus 1/2 to the main mix output * assign a pair of ambient mics to two spare channels on the mixer, and assign those to bus 3/4 only * take the direct bus 3/4 outputs and plug them straight into the line-in of [your] recorder The end result, though this requires careful balancing, is that you can do a stereo or mono mix on the PA, and the same signals, mixed with a bit of the ambient mics, go to the recorder - giving you a balance of the PA and the ambient mics. This usually works quite well, though it can be difficult to achieve a good balance with the ambient mics unless you have good isolating headphones to monitor with and a chance to try things out during the sound check before the show. My stealth recording is done with earworn SP-TFB-2 microphones, so placement is not usually an issue. I have, however, made ad hoc stands for these mics in the past to simplify recording when I'm at a venue where I can get away with leaving the equipment running unattended.

I have yet to have a single installation problem with any version of SS. I started with v2.1 before getting my NH700, and 2.2, 2.3, and 3.0 had no difficulties with installing/upgrading. I have never lost my library or the like, had Db conversion problems, &c. I have experienced a few minor SS crashes and in version before 2.3 I lost several uploads from my own recordings [which caused me no harm as I had backed them up with total recorder beforehand]. I hate to take Sony's attitude on this, but the majority of users do not have the problems you've described, which can only suggest that there is something missing or damaged on your system in particular. I wish I could be more helpful than that, but unfortunately I have never experienced any of the problems you have. There were installer version of 2.3 kicking around before, so perhaps you could try searching the forum for the users who posted those links. I will continue advising that people upgrade, in any case. The changes in speed and stability are worth it for those of us whose systems are not mysteriously incompatible. [That's not a hit on you, either - it's simply the truth. We don't know why your system won't upgrade smoothly.]

Thank you for reporting this, Ryzir.

You know, this could be included in the full-installer versions of SS we link to.. automatically there on install.

PCM is the only uncompressed mode that would allow you to copy without incurring another generation of lossy compression. That is assuming you have the means to copy digitally, and if you're using two portables, you don't. My suggestion would be, since the NH900 is fully backward-compatible with legacy / MDLP discs, to simply continue using your MDs as they are.

When recording I use my NH700 [and have used all previous MDs I've borrowed or rented] only for the initial recording and then playback/dumping the "footage". I never edit on the units, never title on the units, &c. While I find the possibility useful, I have never actually used any of the editing features except for inserting/deleting trackmarks post-recording, and that was a long time ago, before I was using HiMD. This is not to say that I find the features useless or anything. If anything, their presence is part of what attracts me to the format, even if I don't use them. The simple truth is that I have never encountered a need to use the editing features. This is largely because of how I record and edit, though - in single, large chunks, with all editing done in software after the tracks have been erased from their original disc. I do not keep original recordings on their original discs. In fact, I consider the contents of every disc I own to be no more than temporary. For these reasons, and because I never record in HiLP, I would never encounter the problem you have discovered.

The price difference in CAD between the 910 and the RH10 is INSANE. If kurisu's USD MSRPs are correct, then the CAD ones are not. Prices are usually much closer to the exchange rate, and these, again, are not. [The MSRP is out by at least $30-35CAD, btw]

Neither MP3 nor AAC are open standards. They both require liscensing to use, same as ATRAC/3/plus; the difference is that the MPEG offerings are ratified as international standards by a committee [the Moving Picture Experts Group] after being created by various corporations, whereas ATRAC/3/plus is ratified only by a single corporation. OGG+Vorbis are currently the only widespread open standards I am aware of for audio containers and compression. [OGG as a container format, Vorbis as an audio compression codec.] See here: MP3 licensing AAC licensing Vorbis licensing While you're there, this little tidbit is interesting too: Fraunhofer/Thomson patents relating to ATRAC 3 This is not the fault of ATRAC. ATRAC/3/plus are simply audio codecs, and are not actually dependent on any particular medium [including MD, HiMD, and SDDS film soundtracks], just as with any of the other codecs mentioned here. The data is simply data. The container it comes in is another matter. Sony have their commercial and corporate interests to protect, which is the real reason why you can't just manipulate ATRAC/3/plus tracks as files in the same way that you can with MP3 and the like: Sony control both the container format and the stream format. I would assert, since Sony's target demographic for netMD and HiMD are people who want portable music players, that the majority of those people don't CARE what the codec is, as long as they can get their music from their CD to the computer to their player. The codec is entirely irrelevant to them. btw, in case it wasn't obvious, I'm agreeing with you. It's when people realise the limitations imposed by Sony's DRM system [which has absolutely nothing to do with ATRAC/3/plus] that they get miffed and return the players. Again, AAC is not an open codec. Those companies pay for the right to make their own compliant codecs for the format. Again, I agree with you, but given Sony's conflicts of interest [between their electronics and entertainment divisions] and the increasing use of legislation by governments around the world to enforce the use of copyright-protection schemes, this is highly unlikely to happen - UNLESS the majority of consumers suddenly stand up and say, to both their government and the corporations making the equipment, that, "This is crap and we're not going to put up with it." While I'd like to see that happen, it's doubtful that it ever will. People don't stand up for themselves and their fellows on far more important issues [such as human rights], so why would they with something that, in the end, is as insignificant as this?

This isn't the first time we've had conflicting reports on this. In any case, it's strange [at best] that some should experience one behaviour and others something different. I'll try this out and see what it does on my NH700.

Things I'd like to ask Sony: * What is the ADSR envelope of the AGC's normal and loud music modes, and what are their thresholds, assuming -10dBV equipment is used? [i.e. what are the threshold, attack, sustain, decay, and release parameters of the AGC's compression/limiting in both modes?] * What is the resoultion of the input ADC, and if it is a 1-bit converter, is its PCM conversion capable of putting out >16-bit data? * Is the atrac/3/plus LSI capable of directly encoding >16-bit streams from either the internal ADC or optical input without requantising to 16-bit? * Does HiMD employ watermarking? * Are recording levels pre- or post-gain when using the mic preamp? [they appear to be post-gain] * Assuming record levesl are post-gain, what is the "unity" setting? [no attenuation, no added gain to the mic preamp's output] * What are the actual gain levels of the mic preamp in both low and high sensitivity modes, before level control takes place? * Is there any analogue pre- or digital post-ADC lowpass or bandpass filtering on the analogue inputs? [there doesn't appear to be, as demonstrated by audible harmonics recorded on the units when the input signal is >22,050Hz] Many of these depend on what reference standards are used, so I would ask that Sony also include what standard they're using for measurement. I've got more, I just need to think of them.

Independent testing is always useful, though I would note: Sync'ing the output of lossly codecs is suspect at best. Because there is a conversion to/from a completely different domain, timing is difficult to maintain at best. i.e. Maintaining sample accuracy between the original and the encoded/decoded signal is next to impossible, which can account for the differences between the mp3plus* and atrac/3/plus files as much as the differences in the codecs themselves can. Using a more complex signal [such as the actual beginning of a music track] is highly inaccurate, by my experience. The most accurate way I've ever found to maintain the sync of the preface the test track with a burst of a single tone [i.e. -6dBfs 880Hz for 3 secs] and to use to last sample of the burst as the reference point for aligning all further samples. Because of how lossy compression works, the simple tone should still maintain close enough to sample accuracy to be a usable reference. * MP3plus - to my knowledge, MP3plus only takes effect at sub-112kbps bitrates; "plus" is a non-conformant [i.e. non MPEG] extension to the MP3 spec which allows added info about the high end to be included in subcode that non-MP3plus decoders will ignore. Hence, there should be no such thing as MP3plus 256kbps.

I believe that most mic preamp inputs are made for 600ohm impedance. There are many DIY kits and instructions available on the net that give excellent advice on what components to use for battboxes and the like. I'd suggest googling 'DIY battery box" or something.

The truth is that Sony market netMD and HiMD as portable players, not as recorders. The majority of the market they're aiming at don't know that lossless compression exists, let alone how it works, and most of them also don't care. Given the option, most of them would probably never use it [just as they'll never use PCM] as they increase in quality is not worth the loss in capacity [i.e. recording time, not physical capacity for data]. Remember: Sony are marketing these units to a demographic that they think HiLP 64kbps or 48kbps will be of sufficiently quality for. Try telling that same demographic that 700-900kbps [average rates for stereo lossless] is better quality, and they're likely to laugh at you.

Many sound boards have tape outputs, in which case a standard dual-RCA to 3.5mm stereo cable will work fine, even if things are mixed in mono. The AC adapter usually causes problems only when using condenser mics which are powered by the MD/HiMD itself. It should have no ill effect on line-level recordings unless the power lines are extremely noisy.

This might be a confusing answer, because I simply don't know, haven't tested this, and have not heard of anyone else testing it. It's possible that backing up your entire hd using ghost or the like might maintain your DRM info, since all the relevant places forwhat it needs to work right [the windows registry, ss's drm library and settings files, &c.] are all backed up together. I would NOT trust this without testing it, myself. If you have the time, backing up your library to another drive with the SS tool and then restoring your ghost backup would be the [safe] way to find out.

What kind of PC are you using? On my athlon 2500+ it takes about 4-5 mins to rip and encode a 74-minute CD.

Bitrate vs. quality is not a linear relationship. Among other things to take into account are atrac3plus's higher-resolution transforms, which significantly alter what kind of artifacting the encoding/decoding process will manifest. Whether one is better than the other is more a matter of personal opinion than measurable differences, though. To restate this in a different way: If there were an atrac3plus bitrate of 132kbps, same as standard atrac3's LP2, it would likely be of higher perceivable quality than LP2. In the end, this doesn't amount to much of use. While the quality relationship is not linear, it is at least possible to say that each higher bitrate is better in perceivable quality than the next lower one. Mind you, this falls apart at rates below LP3 [atrac3 105kbps]. i.e. atrac3plus 64kbps is likely to be found to be better in perceivable quality than atrac3 66kbps. Clear as mud.

Auto trackmarking overrides the default "mark when silent" mode, apparently.

There are answer to these questions all over the forum already. Yes, you can record your own performances with your NH900; good places to look for mics are http://www.reactivesounds.com and http://www.soundprofessionals.com . There are many threads about mic recommendations in the recording forum. You would also benefit from browsing the HiMD FAQs/essential info forum, a subforum of the HiMD forum, for information about uploading your recordings and such. Assuming that it is an actual MD and not a HiMD-formatted disc from a newer recorder like your own, copying your friend's MD to your computer can be done by analogue means. There are threads in the netMD/MDLP forum pertaining to this. This forum is maintained by volunteers. Expecting an answer within 8 hours of your post is unreasonable. Your patience would be appreciated. Cheers, d.

To the surprise of many, I agree with you on this completely. My own priorities are such that .. well, in all honesty, to achieve the same quality as PCM on HiMD with analogue equipment, you'd be spending a few thousand dollars on equipment rather than a few hundred.

The last time I checked [which was long ago, with SS v2.2] SS's encoders were still not quite as good as the hardware encoder on the units themselves. Since then, a few users have retested this and found that, at the least, the lower-bitrate modes encoded with SS 2.3 and 3.0 are significantly better than they were in previous versions. One test, on hydrogenaudio I think [though the tester did not specify what their recording chain was, making the results at least slightly questionable], came up with the result that the SS 2.3 codec was in fact slightly better with HiSP than the hardware LSI was. In other words, no one really knows, it's all a matter of opinion, and considering the marginal [at best] difference, you're better off using Simple Burner or SS to do your conversions and save the time you'd otherwise be using waiting for realtime transfers to finish.

Keep in mind that I'm dealing with theory, though I'd say fairly plausible theory. Go ahead, A440, and blast me for dealing in theory again. ATRAC SP since v.3 on decks has supported direct SP encoding with at least 20-bit PCM audio. To my knowledge, ATRAC has gone on to support direct 24-bit conversion in later incarnations. I am making the assumption that in more recent years, the same has been extended to all ATRaC/3/plus modes, or at the very least, HiSP on HiMD, since the higher-resolution input stream wouldn't really benefit lower-bitrate modes]. If I'm wrong on this, I'd appreciate someone pointing me to an info source that can correct me on it. The 16-bit 44.1kHz limitation on PCM mode is likely there for several reasons: 1. Because it's a well-established standard that meets basic recording requirements 2. to limit the quality of the recordings to be less than that of professional equipment [if you think this is specious, do some reading about DAT and why there are 2 different 32kHz modes in addition to 44.1kHz, which was locked out on most consumer equipment, and 48kHz, which was considered "analogue" mode.] 3. to limit the quality of illegally copied, copyrighted recordings by more or less incurring a generation loss [assuming sources of >16-bit material are available] 4. to keep the recording length of the media reasonable 5. to lower power requirements [a lower bitrate means slower buffering, lower disc rotation speeds, lower write speeds, &c. all of which affect power consumption] With consumer equipment, ADC conversion is generally done with 1-bit ADCs simply because they are cheaper. They are also scalable in the sense that you can convert the 1-bit stream to whatever sampling rate and bit depth you like, as long as your "chipset" has the power to do the math. [This is remarkably similar to how DSD works, incidentally.] It's a pretty safe assumption that our portables all use 1-bit ADCs. The bit-depth of the PCM output is pretty much arbitrary, to my understanding. If your encoder can deal with >16-bit streams, why not use them? The limitations imposed on PCM recording are mostly practical. Using a higher resolution source stream for lossy modes doesn't use more space, so it's not impractical as long as the initial conversion to a set PCM bit-depth is capable of dealing with >16-bit data. My apologies for being so far off-topic.

HiMD has no crossfade mode [what you're talking about]; neither does SonicStage.

Yes. Read above.