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I hate it when /. removes my anchor tags... http://users.belgacom.net/bn967347/

This has been out for a while (last april):http://users.belgacom.net/bn967347/

Here's another explanation: http://www.aaronrogers.com/nintendods/wifime.php

and also of DS interest: http://ds.darkain.com/hack/

The DS has already been somewhat hacked: http://users.belgacom.net/bn967347/

And not only that, but with a GBA flash cart and appropriate wifi card or $20 dongle, you can actually program the thing yourself!

1. one that charges by the minute, for about 2.55 euro/hour (or about 2.55 dollars normally, but since it's value dropped 3.55 dollars)
2. You can also get the other account wich is 'unlimited for about 20 euro's but (and this is the catch)you get charged for every megabyte over 250mb/MONTH. This is limited to 10 euro's so your max bill will be 30 euro. But it's tsill not intersting since it's only Max. 512 Kbps downstream, Max. 256 Kbps upstream, so no good.

The ideographs have to be learnt by rote, since they contain no phonetic information as an aid to pronunciation.

So the short of this is that most Chinese characters can be reduced into smaller component parts (albeit fewer and more complex parts than a word written in an alphabetic script), and often one of those component parts says something about the pronunciation.

Japanese is trickier, because Japanese generally adopted Chinese characters and character-combinations for Japanese words that mean the same thing, so the phonetic doesn't have any relationship to the pronunciation of the Japanese word. (A lot of Chinese words were also borrowed, and in those cases the phonetic still has a connection to the pronunciation. To make it more complicated, lots of Chinese characters occur in both native Japanese words and words borrowed from Chinese, and are pronounced differently in the two cases.)

For more on the composition of Chinese characters (although not much about phonetics), see http://my.execpc.com/~mbosley/main.html . For an example of three common characters with the same phonetic but different radicals, see http://users.belgacom.net/chardic/writing.html and scroll about a quarter of the way down the page to "Another way to obtain a new meaning".

Then explain why they don't sound better?

Explain why modern CDs rarely drop below -15db on the level mdeter?

Answer: because they're range compressed.

I'm not alone in noticing this. Check out here for commentary of others that have noticed it.

Modern discs ARE noticably louder and they ARE range-compressed. And the quality sucks compared to what the technology is capable of.

OK, so it's all very well that you can now use SACD with more accurate signal reproduction, or even DVD-A (isn't that a term used in porn movies? So I've heard) if you want better quality.

Whose ears are actually good enough to listen to 24-bit audio and tell the difference between that and 16-bit anyway? I have often heard it said that analogue transmission of audio is far worse than digital. I don't entirely agree with that, but supposing it's true - surely the cables between SACD player and amplifier, amplifier and speakers are going to withdraw a lot of the benefits of the more accurate signal?

Yes, we can only hear about 20-bit accuracy. The point of the additional accuracy is, therefore, questionable. The difference in quality it will make is miniscule. The LSB on 16-bit audio represents a variation of 0.0015% in the output signal. The LSB on 24-bit audio represents a variation of 0.000006% of the output signal. Can you hear that final bit? Does it make all the difference? Er, no.

Those who say that the MP3 format is too lossy for them might be interested to know that audiophiles can't actually hear the difference between 256kbps MP3 and the original CD recording. Those who think they need still more quality should perhaps check out the MAD plugin which has the ability to decode mp3s to 24-bit, recreating bits that weren't even there in order to improve quality.

As regards introducing watermarks as a kind of copy protection - well, that's just reducing the quality of the audio, which defeats the point of what you were trying to achieve in the first place.

Listening to that on one of those teenie Mac speakers, of COURSE it's not going to sound any different. ;) While I can't entirely vouch for the quality of AAC, from what I've heard, it's really fitting to be alongside MPEG-4 video -- MPEG-4 video was designed for Internet viewing, and IIRC, AAC was designed for the same purpose.

Really, it comes down to this:

In the beginning, there was MPEG-1 and MPEG-1 Layer 1,2,3 audio. Big deal, koz they were all more-or-less firsts in semi-quality video compression.
Then came (in no particular order) MPEG-2 and MPEG-4. MPEG-2 was a format designed for DVDs and other high-resolution media, and AC3, its companion, designed for high-quality audio. MPEG-4 was designed for Internet viewing, and, as I said, AAC was (is) its companion.

That said, there is a lot of crossing over within the formats. Personally, I encode DivX video with MPEG-1 Layer 3 audio (VBR, which is against the AVI standard, bad me, but oh well).

I don't mean to get off on a rant here, but the largest problem I see with MP3 is that people aren't using VBR. It's an EXCELLENT thing: take bits from places where they aren't needed and put them where more bandwidth is needed. You can have a "CD-quality" MP3 file using VBR -- at lower filesizes -- that you would need 320Kbps CBR to achieve.
Well, that, and people are still using Xing. stop it! Use LAME + EAC to rip and encode your CDs. Honestly.

And now, the obligatory plug. For more information on MP3 encoding, visit r3mix.net. Of course, these are facts, not opinions; I couldn't be wrong.

Convert a 256kbps mp3 (lame codec) to wav and burn the that wav and the original onto a cd. Unless you're an audiophile with incredible equipment, I HIGHLY doubt you will be able to do better than random guessing. (eg. get better than a standard dev away from 50-50.)

Before all you audiophiles flame me, go try it and have a friend test you. (And no looking at which song is which before you test. It's easy as hell to bogusly justify a decision if you know the answer beforehand.) Even on nice equipment, I doubt you will be able to tell a difference.

This is a fairly well documented fact :

This is loose reasoning to be sure, but the differences between cd quality and whatever this new standard is are going to be FAR MORE SUBTLE than the differences between cd and 256 kbps mp3. Selling this new standard based upon "higher quality" will be a complete fallacy even IF YOU ARE an insane audiophile.

3. CD doesn't have a low enough signal to noise ratio. The new DVD super audio is a huge improvement.

Reality check: CD was invented to be perfect sound without waste. The 90db signal to noise and dynamic range provides a noise floor that is lower than you can get from any analog source in the recording studio today. The air current in the room of the recording studio is louder than the noise floor on CD. When you use ANY microphone, you will pick up the room air noise. This means that CD already does a better job than we need it to. I already run into problems where CDs can record sounds too loud for analog equipment to safely amplify. If DVD audio is to be believed, then you could record a dynamic range wide enough to capture a jet engine's loudness. This is not possible to reproduce on current analog equipment without distortion and serious damage to your hearing. Again, CD is perfect. Current recordings on CD barely use any dynamic range. Most modern music has a "compressed" dynamic range. Constantly loud and rarely uses a sound below -15db on the level meter. This is a mastering problem. The mastering engineers master modern music for radio play to get their song louder than their competitors so people will pay attention when their song comes on. Take any 1980's or early 1990's CD and put it in your CD player, then listen to the volume. Now take a modern rock or pop music CD and play it. The volume of the modern music is always near or at the MAXIMUM peak level possible. The dynamic range squeezed out. Now, simply put in the older 1980's or early 1990's CD and turn up the volume on your stereo. You'll notice how much BETTER the older recordings sound. There is IMPACT in the drums. Details in the sound. It's more realisitic sounding overall. The older (but still modern) recordings are easier on your ears at louder volume and seem more natural. This is how the CD medium sounds at its best. Do not listen this way on PC or boombox speakers. You need a decent stereo or good headphones to hear the difference.

Found: r3mix.net

To sum up: ID3v2 is a nasty hack, that should have been aborted at birth. It's almost impossible to correctly remove if the tag is damaged, is insanely complicated, and is badly supported. ID3v1 is limited, but fairly well defined and fairly well supported.

The r3mix comparison shows that LAME (and by extension, notlame) is probably the best way to go for mp3 encoding. Some discussion is also provided regarding the advantages of using VBR for encoding in general (click on the "encoding" tab).

As for my own observations, I found that using this set of command line arguments works best:

notlame -v -V 1 -B 320 -p tempcdda.wav [outputfile]

The results are files that will go as high as 320Kbps if needed (on average, they tend to float around 200Kbps) at 44.1Khz. There is no loss of quality that I can hear; the parts of the music that need a full 320kbps get it, and the silent parts drop to 32kbps.

And the players I've tried all seem to work fine; winamp and xmms (using mpg123) have no trouble with the VBR files. I don't know about portable players. For streaming, I use liveice to resample the files to a particular CBR (128Kbps for my apartment, 64Kbps for sending out over the network); so I can listen to the files at whatever quality I want without sacrificing the quality of my archival files.

So, if I did own a Rio or a CD player that would read mp3 CD's, I could just resample the files I wanted to a particular bitrate (pretty much on the fly).

I see no reason why anybody would only sample their files at 160kbps or 256kbps; every file I've done has needed 320kbps at some point to get the best fit. And the size isn't unreasonable; a typical album is between 70-80MB, with some extreme cases taking up to 110MB. Sure, it's big, but I've only used 25% of the 30GB drive I bought for the purpose (which right now costs very little), including the RedHat install to run the machine.

As for using Ogg, I looked into it. If I could be convinced that the quality would be good enough to archive my entire collection, I would have used it. But it just didn't seem mature enough at the time I began all of this (such as no support for joint stereo), so I chose mp3.

Of course, with storage at its current price, I am tempted to give up compression altogether and just store entire CDDA images. A typical album would then take around 700MB to store, but a $200 drive could then be used to keep over 100 CDs with absolutely no loss in quality.

The r3mix comparison shows that LAME (and by extension, notlame) is probably the best way to go for mp3 encoding. Some discussion is also provided regarding the advantages of using VBR for encoding in general (click on the "encoding" tab).

As for my own observations, I found that using this set of command line arguments works best:

notlame -v -V 1 -B 320 -p tempcdda.wav [outputfile]

The results are files that will go as high as 320Kbps if needed (on average, they tend to float around 200Kbps) at 44.1Khz. There is no loss of quality that I can hear; the parts of the music that need a full 320kbps get it, and the silent parts drop to 32kbps.

And the players I've tried all seem to work fine; winamp and xmms (using mpg123) have no trouble with the VBR files. I don't know about portable players. For streaming, I use liveice to resample the files to a particular CBR (128Kbps for my apartment, 64Kbps for sending out over the network); so I can listen to the files at whatever quality I want without sacrificing the quality of my archival files.

So, if I did own a Rio or a CD player that would read mp3 CD's, I could just resample the files I wanted to a particular bitrate (pretty much on the fly).

I see no reason why anybody would only sample their files at 160kbps or 256kbps; every file I've done has needed 320kbps at some point to get the best fit. And the size isn't unreasonable; a typical album is between 70-80MB, with some extreme cases taking up to 110MB. Sure, it's big, but I've only used 25% of the 30GB drive I bought for the purpose (which right now costs very little), including the RedHat install to run the machine.

As for using Ogg, I looked into it. If I could be convinced that the quality would be good enough to archive my entire collection, I would have used it. But it just didn't seem mature enough at the time I began all of this (such as no support for joint stereo), so I chose mp3.

Of course, with storage at its current price, I am tempted to give up compression altogether and just store entire CDDA images. A typical album would then take around 700MB to store, but a $200 drive could then be used to keep over 100 CDs with absolutely no loss in quality.

If you really want to do reliable tests on wav files, then visit PCABX to get the PCABX program and to read more about the testing methodology. The program takes in two wav files, and then chooses one of the two randomly and lets the user decide which of the two is the one chosen randomly. Basically, once this done a good number of times (say, 20) the program can then tell whether the user can actually tell the difference between the two files.

Also, a wonderful website dedicated to the task of creating archival quality encoded audio (which is indistinguishable from the original) is r3mix. Lame even has an optimized parameter that comes from the work at the site, --r3mix! This VBR parameter gives incredible quality at a fairly low bitrate. Check out too a listening test carried out at r3mix that showed the blind preferences of 42 users over a month of time.

>A P90 barely has enough power to decode MP3 anyway.

It has plenty enough power. I had mpg123 playing 256 kbits MP3s on a P60 with only EISA slots, over NFS on an ISA soundcard and 3com Etherlink III card! Not a single skip, unless I started doing anything else on the machine. (The machine was an original Compaq Prosignia or something like that).

>you can hear all those fans and platter spinning around

Rip out the HDDs, boot of the network. Rip the fan off the CPU, and use a bigger heatsink (this is fine for virtually all the original pentium series. You can easily build a power supply without fans. Problems solved :-)

Don't forget to put the server in another room/closet.

>Well, MP3 ruins the listening experience so maybe it's a wash

Check out www.r3mix.net if you are having trouble with MP3 quality. You simply aren't encoding at a good enough bitrate. If you can tell 256 kbps MP3 from the original then you must be a bat.

• General PDF information.
• Comparison of PDF and other formats, including PostScript.
• The PDF spec, from Adobe's website (if you're not boycotting that as well).

• General PDF information.
• Comparison of PDF and other formats, including PostScript.
• The PDF spec, from Adobe's website (if you're not boycotting that as well).

• PostScript hits the market ~1982
• Project Athena announced 1983
• Project Athena starts 1984
• X (X1) released June 19, 1984
• NeWS is released ~1985
• X11R1 released September 15, 1987
• Be evaluates NeWS as the windowing system for their intended OS in 1991

Sun Microsystems' James Gosling created a displayed PostScript as the basis for NeWS around 1985. This implementation was never particularly Adobe/Apple-PostScript compatible and was only licensed from Adobe shortly before Sun abandoned it. However it was the first use of PostScript for a windowing system.

NeXT then licensed & underwrote development of PostScript into Display PostScript (no direct relation to displayed PostScript.) This was the basis for NeXT's NextStep interface and lives on today in GNUstep.

Apple has recently independantly implemented the PostScript-derived PDF from public specifications for it's Quartz rendering layer in it's recently released MacOS X.

Thus you've a single well known, well documented language that's been used for three independant windowing systems over the course of 15 years, two of them independant of the language's licensors. Add that to it's direct application to printing and it's a pretty powerful argument for further consideration as an X-Window alternative/successor.

And anyway, vinyl sounds better.

No it doesn't. Click here then click Myths (can't link directly as it uses a JS redirect to the frames page).

This unit can't even hold 1/10th of my mp3 music collection. 6GB isn't that big, but it is a nice size for a portable collection. (I don't listen to much classical or jazz when driving the car, for example). The fact that these things look like they can be hacked to hold other laptop HDs makes them more promising in my eyes.

I have 80GB. and increasing as we speak. All my files are encoded at 256k for archival purposes. I use Lame -b 256 -ms -h -p and then I don't need to worry about crappy sounding mp3s. Plus, hard drives are so cheap, you might as well encode as best as you can once and not worry about it.

What I want to know is if any of these portable mp3 players play 256k mp3s, since I don't want to downsample my already-encoded mp3s due to further loss in quality.

I'll agree that perception is what matters. However, what souds great on my $48 Labtec speakers at work sounds like crap on my $500 studio headphones at home. The fact of the matter is, most people don't have $25,000 of audio equipment nor sufficiently trained ears to tell the difference. I'll readily use LAME encoded stuff from people I trust, but cringe in horror when I listen to the rapage that Xing's encoder performs to the quality of complex music.

Think of it this way: most people are arguing which color of crap tastes better. Sites like this one and the one in the article are trying to point out that you don't have to eat crap.

hymie

I'll agree that perception is what matters. However, what souds great on my $48 Labtec speakers at work sounds like crap on my $500 studio headphones at home. The fact of the matter is, most people don't have $25,000 of audio equipment nor sufficiently trained ears to tell the difference. I'll readily use LAME encoded stuff from people I trust, but cringe in horror when I listen to the rapage that Xing's encoder performs to the quality of complex music.

Think of it this way: most people are arguing which color of crap tastes better. Sites like this one and the one in the article are trying to point out that you don't have to eat crap.

hymie

Well, actually, there is a reason: the Xing encoder blows chunks. Sure, it's fast, but the sound quality sucks. If all you're encoding is Teeny Bopper of the Week music, then you're not missing out on anything. If you're encoding stuff that's a lot more complex, you're better off with soemthing that doesn't sacrifice quality for speed..

hymie

"I agree with you that mp3s just don't sound the same, regardless of bitrate ..."

If you can read German, read this article from c't magazine which claims that 256 cbr is approximate to CD quality on a professional speaker system.

I got it from this page, which goes into depth on how to properly encode MP3s for near-CD quality. The entire site contains lots of information about encoding, ripping and MP3 quality in general. I'm not claiming it's all true; much of it is quite subjective. But I would cite the c't test as a definitive response to the oft-held belief that MP3s suck for quality no matter what. Granted, you won't find many LAME VBR encoded MP3s on Napster, but that's not to say that Napster or MP3s are worthless. If Napster users really wanted that quality, they could have it; this only shows how little that quality matters to most people.

--

I was under this impression...

Respectfully, I believe you've been blinded by your $200 licensing fee. The Lame versions sound much better than Frau at 128 and beyond. It is possible that your version of Frau isn't equivalent to the one displayed but I find it hard to believe that Frau has improved to a point where their 256k version can compete with the 128k version of lame.

The tools used to create these are readily available, and I'd love for you to run these tests and post the information on the web. Hell, I'd like an e-mail

miracle@nospammage.procyon.com

Respectfully, I believe you've been blinded by your $200 licensing fee. The Lame versions sound much better than Frau at 128 and beyond. It is possible that your version of Frau isn't equivalent to the one displayed but I find it hard to believe that Frau has improved to a point where their 256k version can compete with the 128k version of lame.

The tools used to create these are readily available, and I'd love for you to run these tests and post the information on the web. Hell, I'd like an e-mail

miracle@nospammage.procyon.com

Why use the proprietary Frau encoder when Lame has been proven to be not only faster, but of better
quality?

And regardless of where mp3 ends up legally, Ogg Vorbis will replace it if licensing becomes a huge issue.

go to http://users.belgacom.net/gc247244/q uality.htm if you honestly believe that.
if you've got 80+ GB of space to play with, i'm sure you'd be willing to encode at 256+ Kbps or use vbr.

Actually, according to r3mix, MP3 does very well at high frequencies if you encode with LAME.