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Everything posted by dex Otaku
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An interesting side-note after further reading: it seems that the people who did the work for FhG on what became the MPEG codecs were a part of the MPEG group. So - while I'd still say that MPEG don't actually make anything themselves, it's important to remember that their members do.
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I haven't tried it with FhG removed yet. I expect I would get the same results you did.
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Athlon XP 2500+ Barton.
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(sighs heavily) Look, Mr. Wong. I'm honestly not looking to start a flamewar with you. What I am looking to do is make sure that the information presented on this forum is as accurate as possible, within the limits of my knowledge. That said - if you're so interested in proving your points, why not try doing some research first? As only a couple of suggestions: Wikipedia has excellent entries on the various MPEG formats and layers, as well as ATRAC; minidisc.org's links and research sections have pointers to plenty of information about the various flavours of ATRAC. Please feel free to do some reading to back up your claims, rather than flippantly misinforming your fellow forum members. Indeed it is. As it is with almost all of the other codecs and compression formats out there. Whether the developers succeed in their goal is another matter. See further down for my thoughts on "sound quality". Categorically incorrect. MP3 came about as a result of combining previous research on perceptual data reduction that was being developed initially as an offering for digital broadcast radio. Their goal was to achieve similar quality at 128kbps as layer II audio [musicam] did at 192kbps. Whether they managed to do this is a matter of personal opinion, but that was their goal. Considering the fact that what became layer II and layer III were developed in the late 80s and early 90s, I think they did an admirable job. Also categorically incorrect. As I said, what became MP3 was originally developed with the intent of being used for digital radio. MPEG adopted and ratified FhG's codec as layer III for MPEG-1 because it fit the group's purposes. The format did not originally have anything specifically to do with either video or MPEG. It's inclusion in the MPEG spec is incidental to its original reasons for being. ATRAC itself was introduced at almost the same time for both MD and as used for the optical film soundtrack format, SDDS. Does that mean ATRAC was made for film soundtracks? No. Further - to the best of my knowledge, MPEG does not work by commisioning encoding formats. MPEG is a group that ratifies standards. They do not make anything, per se. They simply bring together a large group of media engineers to assess formats for inclusion in the standards they propose to the rest of the world. It's up to the rest of the world [or, at least, major media distributors and equipment makers] to make up their own minds whether they want to listen to MPEG or not. If sound quality had never been a concern, they never would have started developing it in the first place. As goes with every other codec. Again, whether they achieve their goal is another matter, but their intent is this: Create an an algorithm to fit a specific sound quality expectation and bandwidth requirement. This need not [and often is not] be attached to any particular physical medium [such as minidisc, CD, DVD, &c.] or other information type [such as film or video]. I already stated what their goal was, so I won't repeat it. The other half of this is the use of "sound quality". Particular compression formats are usually made to compromise between sound quality and bandwidth; part of the development cycle is to figure out what kind of quality is their actual goal. The original ATRAC at 292kbps was not significantly better than either layer II or layer III audio [at their target bitrates for interleaving with video], and the goal for all three was as I stated above: to achieve the desired quality within a particular bandwidth requirement. I would assert [though this is purely opinion] that ATRAC, MP2, and MP3 all originally had very close to the same goals in terms of *desired* fidelity. In the case of both MP3 and ATRAC, their encoding algorithms have been tuned and retuned over the years to improve quality quite dramatically. I would agree that ATRAC at 292kbps exceeds MP3 at 128kbps, but that's to be expected considering the limitations imposed on both [by their decoders] and the large difference in bandwidth. Many would argue that today's MP3 encoders at 256kbps or greater are better than ATRAC, however. This is a matter of opinion, though. Back to your semantic argument, I see. Technically, as I already digressed in the previous reply, to 'hear' is for sound to be transduced by your ear and received by your brain. Whether your brain is able to perceive every part of the sound is another matter, hence my using 'perceived' back there. Most people do not make that distinction about the word 'hear' though. That is why I used the word in the first place. There is a reason they are both called perceptual coders, but I've already stated what that was, so I'll let it rest now.
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Does this hopefully imply something such as Sony having listened to their customers, and made some changes that have been a long time coming in many instances?
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To whoever the dedicated reader at 3:38am central time was, there were errors in the graphs which I have now fixed [4:12am].
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Please consider using 'edit' to add to your previous posts in the future.
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This is a spinoff from the thread: Himd Destination Codes I have a NH700. I noted that: .. and decided to test what the difference was between each setting. My suspicion was that this setting is actually just a ratio or threshold setting for the headphone amp. It appears I'm correct. The test... I used two test files: one was a 15 second linear sweep tone from below 20Hz to 22.05kHz, created using the simple synthesis tool in Sound Forge. The other file was 10 seconds of white noise. I wrote these to a CD image and put them on a HiMD-formatted disc using Simple Burner in PCM mode. The peak level of both files was -3dBfs. --> graph of the sweep tone --> graph of the white noise file I recorded a reference WAV file for comparison by looping my Revo 7.1's output to its input, running things at full-volume, and leaving the record levels set to unity gain. --> graph of the reference [loopback] sweep I set my NH700 to each mode listed above and recorded the test tracks on the computer three times: once with the volume set to 20/30, once at 25/30, and once at 30/30. I also repeated this twice more, with the setting set at '00' and 'FF', to see what the differences would be. I then repeated all of the above with the EQ set to the UNIQUE setting. [Note that things were not actually done in this order, it's just simpler to present them this way.] Once I had these recordings, I split them up into individual files for analysis, organised by what the setting was, the volume level, whether EQ was enabled, and finally which test file it was. Here are the results for comparison, as graphs.. Linear Sweeps: --> Volume = 20, no EQ --> Volume = 20, EQ = UNIQUE --> Volume = 25, no EQ --> Volume = 25, EQ = UNIQUE --> Volume = 30, no EQ --> Volume = 30, EQ = UNIQUE Whitenoise: --> Volume = 20, no EQ --> Volume = 20, EQ = UNIQUE --> Volume = 25, no EQ --> Volume = 25, EQ = UNIQUE --> Volume = 30, no EQ --> Volume = 30, EQ = UNIQUE In all cases, the various results were overlaid for comparitive purposes. White = FF Green = A0 [Europe/UK] Blue = 25 [Hong Kong/Korea] Magenta = 20 [E91/Mexico/Canada/Australia] Red = 00 I did not alter the scale of the graphs to make differences between every set obvious. In every case, the higher the value of the setting got, the lower the volume got. The difference between 00, 20, and 25 was so small as to be considered measurably negligible. A0 is definitely crippled by comparison. The graphs may be confusing because of the overlap. In almost all cases, 00, 20, and 25 overlap completely; A0 and FF also overlap for the most part, though as the volume setting increases the differences become more pronounced. The effects "crippled mode" have on the EQ are very obvious, to say the least. I noticed while making the graphs that with volume set at 30/30 and the "region" setting set to other than A0 and FF, the EQ would cause distortion in the upper frequencies. For interest's sake, here are 3d graphs to show this: Setting 00, 20, 25, A0, FF. I have left my NH700 set to '00'.
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Many decks have optical outputs, but not all. I would suggest browsing eBay for decks, and using the minidisc.org's equipment browser as a reference for what decks actually have what features. You would need a sound card or USB adapter with an optical input. I don't really have specific suggestions on this, but there are USB adapters for well under $100USD that have optical in- and outputs on them. Copying via optical means is done in realtime, just so you know. Asdie: for making analogue copies, it helps to calibrate your sound card's input. If you have a netMD, you can take a WAV file of a 1kHz tone at 0dBfs [the highest level, digitally speaking] or -1dBfs, put it on an MD, then use that [since you know it's as loud a signal as can possibly be recorded on the MD] to set your recording volume before copying anything. Done properly, this will prevent clipping from ever occurring in your dubs. It was moved because the topic of this thread has absolutely nothing to do with live recording, and has everything to do with you looking for a new MD unit.
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This is a specious semantic argument, but in any case, the point was that ATRAC and mp3 operate on the same basis - data reduction based on perceptual coding - which throws out data that is not likely to be perceived.
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The bit about the manual is my .sig - not directed specifically at you. You can find most MD/HiMD manuals online at http://minidisc.org/part_Manuals.html
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Cool. AAA are anaethema in my world. They are insanely expensive compared to AA.
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Oi. A flash-based device limited to USB 1.1? That's a huge throwback.
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It would also be a great deal faster at transferring tracks from your PC than any netMD or HiMD can be.
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At least 60 years of psychoacoustic studies prove you wrong on this. mp3 and all variations of ATRAC use the same principles to do the same job. Reread my previous reply [above]. Perhaps it's too technical, I don't know, but you're completely incorrect on this.
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Indeed, but only from devices that have digital outputs. MD portables generally do not, and none of the HiMD portables do.
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First - in the future, please consider making your topic's title and description relevant to what you're going to ask. An example in this instance might have been "Which HiMD should I buy?" If you don't need recording capabilites, the NH600D might suit your needs. If you're going to use the unit for recording, don't get the NH600D. That 'D' there means 'Downloader" and that the unit has no recording capabilities at all, i.e. no mic input, no line in/optical. The NH700 is the base model recorder. It does not have a radio tuner - that would be the NHF800. Cheers and have fun with your upgrade.
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Ah. Yes. You need an external preamp. And welcome to the fora.
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All lossy audio compression formats use basically the same set of techniques and theories to do their job. The techniques incorporate established research done on how human hearing works, such as the absolute threshold of hearing and multiple forms of masking [one sound obscuring another]. There are many variations on how to use these theories, and many ways to implement them, such as using different mathematical techniques [mostly variations in discrete transforms/fourier analysis]. In all cases, lossy audio compression works by analysing the incoming signal and basically throwing away parts that are not likely to be missed - the parts people aren't likely to hear anyway. Distortion [artifacts] occur when signals get too complex for the resolution of the encoder, such as when multiple instruments with complex harmonic overtones are playing at the same time. WAV audio is uncompressed PCM. PCM data is not even of the same domain as most lossily-compressed formats; PCM is amplitude vs. time, whereas the majority of lossy formats store data in some variant of amplitude vs. frequency.
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How about doing what everyone has been doing with MD for years? Copy your MDs to your computer via analogue connection. Unless your sound adapter is absolute crap, the loss in doing so [carefully and properly] is pretty close to nil. I'm not fully aware of what the quality is like of most Macs' builtin sound, but I'd presume that Apple couldn't get away with cutting such a serious corner when so many of their customers rely on their equipment for these purposes. It's unfortunate in the extreme that Sony do not support Mac as a platform [at all], however, until the advent of HiMD recordists who were PC users were in the same boat anyway.
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FLAC support has been part of HiMDRenderer for some time now.
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Have you tried updating MDAC? [MS data access components]
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Sony's software is Window-only, however.. MD does not upload [transfer -to- a computer] in ANY case, and HiMD does upload, but has no Mac support. You can of course use etiher with any computer or editing system by transferring the analogue way. If you have a quality sound card/adapter, the loss incurred is really not an issue.