dex Otaku

As much as I've demonised Sony's software division in the past, this really seems like a matter of having half-implemented something and forgotten the other half or just not gotten to it yet.

--- After rereading my own words, I'll just withdraw my utterly pointless argument. I agree with greemachine, who wasn't even talking about what I was arguing pointlessly about. er, greeNmachine. yeah.

I don't dispute that one algorithm may be less processing-intensive [and thereby more energy efficient] than another. What I dispute is that there is some inherent link between perceivable playback or recording quality and low power consumption. IMO, they would likely design a hardware codec to maintain a certain level of quality, then tweak it to lower power consumption. This is also a place where custom-designed circuits or DSPs can make a huge difference. If you know that you have a dedicated atrac/3/plus LSI, you can pretty much count on that consuming less power than a general-purpose DSP with firmware-loaded codecs which wastes a lot of energy on being versatile. This is a positive argument for proprietary design, in fact. Thing is - even general purpose DSPs these days have so much processing power that basic jobs like decoding simple audio should be nothing to them [even to run in many times realtime - SS with software codecs is STUPID; they should be sending digital audio over USB to the recorder and having it encode, with hardware, the audio to be written to the disc]. I would think that buffering and mechanical operations would consume far more power than simple decoding cycles [encoding is another matter], buffering especially. But then, I'm theorising. It's only my opinion, and this time around it's not based on anything I've researched, just what I consider to be common sense - which is as likely to be wrong as not.

My mp3s tend to be of the lame "--alt-preset standard" quality or higher. I find most music to be pretty transparent encoded this way. Likewise for iTunes' FhG encoder set at "VBR highest quality". For some things I use lame "--alt-preset insane" but it really depends on what the material going into the encoder is. For distro material I use "--alt-preset standard" which, frankly, exceeds the quality expectations of the majority of people I've done work for. About 1/5 of my music collection is in WavPack or FLAC, as well. This really depends on your ears, your 'phones, and how you listen. I find that non-hardware encoded LP2 sounds like sand or grit pouring into my ears. The artifacting is very obvious, and very annoying. Hardware-encoded LP2 isn't that bad, though. I'm basically agreed on the bitrates gm has listed, but there's something here I just have to comment on.. Does anyone else out there besides me think that the whole "designed for low power consumption" thing has nothing whatsoever to do with the sound of the hardware or [especially] software codecs? Every time I read someone saying this, it's like the proverbial scraping of nails on a blackboard in the back of my mind. My personal opinion is that there is no relation between these factors whatsoever. Second, MD and HiMD are limited to realtime CBR encoding because they're, well, realtime encoders [i.e. live recorders]that are expected to work a certain way. It's not that VBR encoding on the fly is difficult to implement by itself, it's just that things like - estimating how much time is left on the disc, for example - are made extremely difficult by it. Same with realtime lossless packing. I would also wager that designers for companies like Sony probably think in terms of, "The recordist probably wants a guarantee of a certain quality when recording." This is all beside the point though, because this topic is about software-encoded content downloaded to players from computers, not live recordings. So I'll shut up about it now.

I'd be either updating, or uninstalling/reinstalling my video drivers first off.

I fully expected the new 352kbps atrac3plus rate to not work with Simple Burner [which would require an update itself to show the bitrate, even though the codec is present after installing SS 3.3] .. I am, however, a bit surprised to see that if I try to import a CD into SS to put on HiMD, the 352kbps option is completely missing. MP3 - sure. WMA - sure. Atrac lossless - sure. But no 352kbps in the atrac3plus menu, which maxes out at the [HiMD incompatible] rate of 320kbps. Another surprise .. there is now a menu option in there for "encoding quality" - either "normal (fast)" or "high". It seems some of our requests have been listened to once again, which is really nice to see. Hopefully I'll get around to trying this out with LP2 to see if there's any difference between them. I found LP2 encoding to be absolutely dismal with most audio in previous versions. As an addition - while trying to import tracks with the quality setting on "high" and nothing else producing system load, SS crashes. Side-note: yes, 352kbps is available to me as an option from the library, meaning I can transcode files of any format SS understands to a3+ 352kbps. No CD importing, though.

I stick with HiSP [256kbps] for music. I occasionally use PCM for some things, such as classical or jazz, but not often. Some lo-fi recordings I have used LP2 [132kbps] but I generally find that 256kbps is the minimum bitrate that my ears can stand. And yes, I can hear artifacting in first-generation HiSP-encoded material, though not often enough for it to bother me with portable listening. Note that when importing CDs, 352kbps is not an available option [at least - not here, it's not]. 352kbps is only available when converting files imported into your library from other sources [like WAV files]. If the recent reports from greenmachine et al about AAL are accurate - that the lossless copy is for playback from SS only, and that anything transferred to HiMD at a different bitrate than the "sub" rate selected is encoded from the already-lossy copy, then I see AAL as completely, utterly pointless. I don't use SS to listen to anything except mic recordings I've just uploaded.

oh, and manufacturers..

Taxes are standard, the levy on blank media is less than a dollar. The rest is simply price gouging by retailers.

Hi-MD blanks in Canada .. Minidisc Canada: $8.99+shipping Futureshop: $11.99 TheSource [formerly Radioshack]: $13.99 Futureshop is the first place to actually stock them in the town where I live. They've been open less than 6 months. They don't stock players or recorders. TheSource stocks players, but the only ones I've seen in store [only one store here] were the NH600D and RH710. Wal-Mart also still has some old stock, netMD players. Average price of MD80 blanks here is $10.99 for a package of two. Yes, those prices are freakin' insane.

I started recording sounds with portable cassette recorders as a child. Some of my earliest experimentation was with home dubbing machines and cassettes of the Muppet Show, bits of audio taken from TV, chunks from CD [which had only been around about 2 years at that time], and anything else we could catch bites of with quick use of the pause buttons on my friend's cheapish home stereo. That friend has since gone on to be a DJ of some reknown here in Canada, having been a part of keeping the Winnipeg hiphop scene going through the 90s. [search for "DJ Hunnicut"] My first "portable" recorder was a Technics SV-100 digital audio processor, which matched the size and design of a hip-pack VHS recorder Technics made to go with one of their video cameras. This unit converted analogue audio to 14-bit digital, and then fit that into a standard 29.97fps video signal for recording on any VTR with a composite input. I still have my VHS-based digital recordings, including the first live recordings I made in high school with $4,000 microphones on loan from the local uni's music department. They still play, and the quality, while not astounding, is still quite good. Around the same time I came across articles in my favourite magazines at that point - Stereo Review, Audio, and Stereophile - about DCC and MD. The debate in these and other trade publications about lossy compression encouraged enough heat under my collar that I actually wrote at least one research essay for english class back then about the evils of data reduction, and how the careless use of it would become the wave of the future. MD equipment was seldom-seen in rural Canada back then. There was no marketing for it, but if you dropped in to a Sony Store you could gave at the tiny recorders under the glass that were priced between $1,000-1,500CAD, not that much less than consumer DAT recorders. The first person I met who owned an MD recorder of any kind was back in broadcast college [1994]. I remember asking him what he thought about the data reduction it used - he said, "What data reduction? It records just like CD." I remember getting into a jovial drunken debate about whether it did or not at the student pub, which ended with his running over to the residence and grabbing the manual, and eventually his digressing because, well, I was right. The next time I saw MD was during a contract job for CTV at the 1995 Nordic Games, where CTV hired a number of students from the school I went to. In the radio rooms at school, and even in the TV studio, everything that used sound clips was still using cart tapes. CTV had these nice, professional Sony units that were the same size as a cart machine but used MD - the editor I set up some of that equipment for loved them, since you could cue things so easily, title them clearly, and the sound quality was so far beyond cart tapes that they bought into the format and never looked back. Since then I met a number of others who had home or car decks, but portables were rare beasts, taking until after about 1997 to fall below the $1,000CAD mark for basic recorder units. I had a few friends [such as one of the people I use to work for and with doing live sound, who taught me a great deal of what I know] with recorders, but as I said, they were extremely rare to see. When I attended a private audio engineering college in Vancouver, BC in early 1996, the basics of editing were still taught with 1/4" open-reel tape, though they had one large studio with a Sony 24-track DASH recorder and a few dedicated ProTools stations for advanced editing. No one in the audio eng department used MD, though it was used in embedded applications in the radio and TV departments in the same way that CTV had earlier replaced all their cart equipment. It wasn't until about 1998 or 99 that one of my closest friends bought a portable to use for stringer recording of interviews for CBC. She put it in my hands and let me play with it. I experimented a bit and found that the sound quality was indeed far, far better than my expectations had been [even after having been experimenting with software mp3 encoding since about 1993 - back when it took a day to encode a single 4-minute song]. Her first unit didn't last long, thanks to an incident involving her nephew and a bottle of root beer, and her replacement was a venerable MZ-R37 which I would use a few times over the next several years. In 2003 I worked on an audio presentation for an art gallery installation that meant having to track down or record a number of sound effects [thank goodness for the BBC royalty-free library and R. Murray Schafer]. My producer's contribution to the equipment manifest was his friend's MD recorder [not sure what model it was, possibly something with a 70 in it, the most basic recorder of that model year] and a Sony MS-907 microphone. We spent many an evening tramping around in the grass or waiting for trains to drive by. The quality of every recording I made, despite the bottom-end recorder and the cheap microphone, blew my socks off. Again, my expectations were exceeded every time I brought home footage to log, even having to copy everything via analogue means. In 2003 my ex-lover gave me a lump sum payment for a job we did together at the same time as my tax return came in, and several other friends threw in cash at my birthday party so I could finally get a portable recorder. That was the summer I pre-ordered my NH700 from Minidisc Canada. My decision to go with HiMD was based mostly on the fact that recordings were uploadable, and that it did PCM. Or.. actually, my decision was based mostly on cost. I was aware even then of alternative formats for recording, but the cost of all other formats was so high compared to HiMD that any possible hassles of being an early adopter were easily set aside. I now have both my NH700 and an RH10, thanks to Kurisu. I am extremely happy with HiMD's performance, portability, and usability in terms of recording. I plan to use both units until they die a dignified death, having been well-appreciated. I do however doubt that my next recording equipment purchase will be HiMD-based, whether they continue with the line or not. I fully expect that someone will market an affordable, and more importantly usable, sufficiently-high-quality portable that uses either a hard disc or flash memory to record 24/96 digital audio with no DRM restrictions. Chances are, whatever I buy into next will be professional equipment with XLR inputs, decent mic preamps, and the like. I have absolutely no regrets about having gone with HiMD, or having used MD in the past, though. For the sake of recordists everywhere, I hope the format sticks around for a few years yet, because its affordability more than offsets the potential hassles of DRM or being restricted to using Sony's software.

I made a visit to the local Futureshop, which only recently opened, a few days ago. Walked away with my first Hi-MD blank [for $11.99CAD] other than the one that came with my NH700. I'm quite happy to know there's now a local source of discs for less than $14 a pop, but I have to relay the conversation with their floor rep because it was just .. funny. rep: Can I help you find something sir? me: Yes. Do you sell Minidiscs? rep: Ah yes, follow me. You mean the ones like video cameras use, right? me: No, I mean Minidiscs. rep: [blank look] .. [hopefully] uh, like the kind that video cameras use? me: [getting annoyed] No, I mean Minidiscs. The audio format created by Sony in, oh, 1992? For portable recording? rep: I'm afraid I'm not familiar with that sir, but I'll ask the other guys. I follow him over the the video camera section, not getting my hopes up. rep: Hey, uh, do we sell... me: Minidiscs. 64mm disc, hard plastic shell, been around since 1992? rep 2: Ah yes, they're over.. I look down at the shelf of blank video recorder media we're standing in front of. me: They're right there. I pick up a package of 2 MD80s, priced at $10.99, pretty typical for around here. rep 2: We also have the high-capacity ones here somewhere.. I look over the shelf, and find, to my absolute surprise - single clear-blue HiMD blanks hanging from a hook. me: Oh my god. You actually sell these. Wow. Uh. Thank you for the help finding them. rep 2: You're welcome. [as the first rep slinks off into the background] Heh.

A wink in the original post would have been appropriate. At this point, deleting the thread would probably be more so. My communication skills seem to be lacking. Sorry.

I understood the reasoning before you even replied. It doesn't change my opinion that it's aesthetically displeasing and breaks continuity in reading. I keep in mind that neither really has any actual effect on usability, and I'm not saying I wish you'd get rid of them. I just don't like the way they look. I could have been clearer in the frst place, I know.

Was the recorder on your person when recording, or was it placed securely somewhere? I have had similar things happen when the recorder got bumped. This causes a write error on the disc, which after uploading with SonicStage will exhibit one of several different behaviours, from playing properly but not being uploadable, to not playing properly or being uploadable, to trashing the rest of the disc from the skip onward. Did the problem start after you uploaded the recording?

the link is that I like the sound of a352, even though it reminds me of A/52, which is either [i don't recall clearly at the moment] a synonym for AC3 or the name of a protocol used to carry AC3 over SP/DIF, either way, having something to do with AC3. There is otherwise no connection between AC3 and any form of atrac that I'm aware of.

As I see it, there are three basic kinds of normalising: 1 - and this goes with A440s first listed def above, is to increase the level of a track so that its peak levels are just below 0dBfs [digital maximum]. This is specifically called peak normalisation and does not involve dynamics processing other than a single volume change to meet the new peak level, which occurs without the possibility of clipping distortion because it depends on analysing the entire section to be processed first. There is no compression involved. Almost -no- consumer software, and -no- form of realtime normalisation as performed by directx or VST plugins use this method. 2 - to measure the RMS peak level of the section to be processed and then increase levels to match a specified RMS level, dynamically compressing the signal if levels exceed a certain amplitude. This is specifically called RMS normalisation by most who use it. This is typically the most even-sounding way to do normalisation, though if your levels vary greatly from track-to-track [i.e. mixing 80s or early 90s-mastered CD-sourced music with the current bitpushed garbage that nearly all CDs contain] the results can be less than pleasing, with "newer" tracks falling dramatically in volume and older tracks often still being "too quiet", depending on the exact settings used. This is also offered by software like sound forge, whose "normalisation" dialogue has the option for either peak or RMS normalising. Both methods require analysing the *entire* section to be processed first, and can not be done in real time. 3 - to attempt to match levels "instantaneously" to a specified or pre-set level, usually [again] by RMS processing, though either RMS or peak can be used as well as both in combination. This is simply a form of dynamics compression, and is probably the most common type of "normalisation" used by non-audio professionals. I would go so far as to say that this is what most non-pros are referring to when they say "normalising," and is, IMO, the absolute least-deserving method of the title. And yes, I'm using the term 'pro' pretty loosely. This is basically a post-production equivalent to using AGC when recording. It typically involves raising levels through quieter passages to meet a minimum RMS amplitude as well as dynamic peak compression, which lowers levels above a set threshold by a ratio that changes depending on how high the input level is above the threshold, usually maxing out at infinity:1 [brickwall] limiting. This can be done in real time as it relies on flying by the seat of its pants to work in the first place. A good example of the last method there is iTunes' "Sound Check" option, which dynamically compresses all tracks played or burned. iTunes, Nero, and virtually all other burning software that offer an option like this have pretty liberal compression settings which are best-suited to use with all the same genre of music or age of recordings, for the same reason as I mentioned above - the wider the range in what you're assembling as a compilation, the more difficult it is to match everything without making all of it sound like crap. In any case, the last method is the -last- method I choose to use, simply because it tends to not work well. It's often easier spending an extra few minutes to match tracks' overall volumes by ear by jumping back and forth between them than it is to rely on a tool that gives you no access to any of its settings, with the foreknowledge that its processing is purposefully middle-of-the-road. For example - I only use this if I'm making a one-off disposable compilation to listen to on a road trip, when I want to spend the least time possible building the compilation. For anything else I do it manually and carefully, because I'm fussy and I actually do care about how bad the results generally are doing it this way. In a possible move to eliminate some of the confusion, many companies have started renaming their plugins as "maximisers". These are basically optimised bitpushing [limiting] tools. I would hazard to say that purists pooh-pooh most forms of maximising or normalisation because any change in dynamics is basically messing with the source too much to be considered acceptable. My own opinion is that if your source material is fairly well-balanced in terms of levels, there's no need for dynamics processing at all; peak-normalisation is fine as long as it's followed by proper dither at some point [as altering levels digitally in any way always induces a certain amount of error, usually in the form of aliasing distortion]. If your source wander considerably, then RMS normalising is possibly a good idea, if carefully applied with manual settings by someone who knows how compression actually works and can avoid its pitfalls. Lastly, even I use a maximiser [mostly for its high-quality dither as the final processing stage] for most recordings now, because much of what I record involves rather extreme dynamics to begin with. Liberally applied [i.e. so that the total number of bitpushed sections totals hopefully less than about 1-2% of the whole recording] this doesn't do significant damage, and can greatly increase the average level of the recording without damaging the overall dynamics. The key is to know what the specific tool is doing, and not to go overboard. Compression is very easy to abuse, and people tend to forget that there's such a thing as listening fatigue - which compression helps induce faster. Experiment with settings, read up on what compression, compansion, and expansion actually are [A440 lists an excellent resource for this above, and even wikipedia has good stuff on related topics], and of course - use your ears and LISTEN. And lastly - to those who like to bitpush the crap out of their recordings, my suggestion is that they turn up the volume knob instead. It has the same net effect [louder overal levels] without the distortion or limiting, and maintains the dynamic range of the recording rather than destroying it utterly. To sum up: Peak normalising = level change with no compression or clipping [used correctly, at least]] RMS normalising = level change with dynamics processing if peaks exceed 0dBfs [or lower, depending on the algorithm/settings used]; dynamic compression basically = bitpushing if levels are consistently high. "Normalising" as done by most consumer and burning software = the post-pro equivalent of AGC, with recording levels slightly boosted to bring up quiet parts and everything above a certain level compressed with a dynamic ratio which eventually meets brickwall limiting and/or bitpushing. Generally speaking, this will give the worst overall results of any method available, but is the fastest to use since it involves no user-accessible settings or auditioning. Cheers. P.S. if anyone can think of a way for me to use the word "levels" more time in a single sentence than in the above, go for it. As long as it's not complete gibberish, I guess. P.S. if anyone can think of a way for me to use the word "levels" more time in a single sentence than in the above, go for it. As long as it's not complete gibberish, I guess. Oh, and P.P.S. you might consider changing the topic of this thread to "what is normalising" since that's what it's about - relevant topics mean people can find what they're looking for. P.P.P.S. having re-read the original post, scratch that. heh.

One thing to say: Wow, is that adbot thing that inserts its own little posts ever annoying as hell.

It would be nice if they would update Simple Burner to include 352kbps as well.

I don't mind the idea of a352, though it's ATRAC3plus, not ATRAC or ATRAC3 .. my one additional thought on this is that it's similar to A/52, another name for AC3, aka dolby digital. In any case, I think in bitrates, so I will remember it myself simply as 352kbps.

It does indeed work, mgdimo. You can title MD and HiMD tracks with either netMD or HiMD recorders and SonicStage.

Well - I never use HiLP, on one hand. On the other, basically everything I record will end up in editing at some point, so it has to be in a usable format in any editor. No form of OMA meets that, but WAV does meet that requirement. I normally copy to WAV then back up as WavPack. Yes - I used to use FLAC, but after realising the WavPack encoder has basically the same type of corruption-proofing as FLAC [which APE lacks], that it compresses slightly better [2-10% better than FLAC for ambient recordings], and that the encoder is literally about 20 times faster .. I made the switch. I still use FLAC for backing up DAO images since the tools for embedding cuesheets are more readily available than for WavPack at the moment. 2nd note - if multiple compression passes [i.e. successive generation loss] isn't an issue to you, you can always transcode, say, HiSP -> WAV -> MP3 or whatever you like. I only back up lossless formats, myself [not counting the initial pass of encoding as when recording in HiSP].

A few quick notes from over here: * If the camera has manual level controls, go ahead and use it. If it doesn't, use something else that does. It makes all the difference [manual vs. AGC]. * There's nothing wrong with the recording formats supported by DV25. If your final destination is CD, just be aware that you'll be resampling everything recorded on DV. [48 or 32kHz to CD's 44.1kHz] * The mic preamps built into consumer video cameras are extremely dodgy at best. If you want good clean recordings, go with a format dedicated to audio. Even with the limitations of the tiny low-powered preamps in MD or HiMD recorders, they will almost always do a far better job than any consumer video camera, and quite a few prosumer cams as well. * Lastly - if you can meet your own expectations with equipment X, i.e. if it works for you - use it. Our advice is not the end-all be-all. Cheers.

I have tested this thoroughly with a RM-MC35ELK remote [single-line display] with both an RH10 and NH700. In both cases, making recordings using the internal mic preamp is a totally fruitless effort, filters or not. The noise created by the display updating gets picked up by the mic cable by the proximity of the plugs alone. The same problem has not affected line-in recordings I have made with the remote plugged in [mainly due to the input level difference alone].

Putting XLR jacks on the equipment is tantamount to an instant upgrade to "pro" status. It would be nice if they made both versions. [or if a 3rd party would make a HiMD version of something like HHB's MD field recorder] I've seen broadcasters [i.e. the CBC] actually take older metal-cased MD recorders and bolt small steel plates onto them which hold a single XLR connector [no phantom power of course] and is cable run directly to the mic input, too. I would actually suggest, especially for low-level ambient field recording, using an external preamp anyway. Most MD and HiMD recordists use mics whose self-noise is high enough that the mic preamp's noise is basically at about the same level [or lower], but professional mics will outdo the performanceof the built-ins time and time again for either very loud [driven to clipping] or really quiet [buried in the noisefloor] recordings. You might like to look at something like this in terms of stepping up: Marantz PMD-671 I can't attest to the quality of its built-in preamps but featurewise alone it blows HiMD right out of the water. Mind you, it's also a hell of a lot more money. Cheers.