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AnalogWhole, an Alternative To FairUse4WM

Squidmarks writes, "AnalogWhole is a free application that allows any file that can be played in Windows Media Player to be transferred to iTunes as an MP3. It uses, you guessed it, the 'analog hole' to re-record any DRM'ed song as an MP3. Because the analog signal doesn't actually leave the computer, but is simply looped back in the sound card, sound quality of the re-recording is excellent. All meta data is transferred as well. The MP3 file is automagically added to iTunes. Just show it where you store your DRM music and walk away."

168 comments

  1. Sound quality by CRCulver · · Score: 1

    Because the analog signal doesn't actually leave the computer, but is simply looped back in the sound card, sound quality of the re-recording is excellent.

    ...as long as you don't actually try to play the file on anything more than the cheapest and flimsiest stereo and speakers.

    1. Re:Sound quality by kiwimade · · Score: 3, Funny

      Considering its targeting stuff like ipod playback, this shouldn't be a problem.

    2. Re:Sound quality by geeber · · Score: 1

      Because the analog signal doesn't actually leave the computer, but is simply looped back in the sound card, sound quality of the re-recording is excellent.

      Actually the quality of the conversion has little to do with the fact that the signal does not leave the computer and everything to do with the quality of the A/D and D/A converters in the sound card. Given the consumer grade sound cards in many computers I am skeptical of the claims of quality.

    3. Re:Sound quality by Metteyya · · Score: 2, Insightful

      Bullshit. Or, if you like it that way, you're right, but that's completely not applicable here. It's just that signal - still in digital form - is received by another app, that's all. Sort of like JACK works - manages exchange of many (digital) audio "streams" between applications. So it's something completely different than "physical" loopback, like plugging your card's line-out to its line-in. Some audio apps already work that way (mentioned JACK for Linux, for example), the only new thing here is automatisation of the whole process and using already available players in the system.

    4. Re:Sound quality by cheater512 · · Score: 1

      If your a audiophile then wtf are you doing buying DRMed music?
      Wouldnt you be ripping CDs?

    5. Re:Sound quality by drgonzo59 · · Score: 1
      Bullshit. If any A/D or D/A conversion occurs inside the PC case there will be noise (and lots of it -- according to professional standards).

      That is why professionals never use internal sound cards for A/D (yes, Creative is considered crap). For a more serious option check out this baby from Roland : ahref=http://www.rolandus.com/products/productdeta ils.aspx?ObjectId=758&ParentId=114rel=url2html-241 56http://www.rolandus.com/products/productdetails. aspx?ObjectId=758&ParentId=114>

    6. Re:Sound quality by CastrTroy · · Score: 1

      I don't think professionals use regular PCs either, So I don't think it reall makes a difference which sound card you get. People who buy sound card like that are the same people who buy equipment like this

      --

      Anthropic principle: We see the universe the way it is because if it were different we would not be here to see it.
    7. Re:Sound quality by uvajed_ekil · · Score: 1
      Or, if you like it that way, you're right, but that's completely not applicable here. It's just that signal - still in digital form - is received by another app, that's all.

      So the signal is "intercepted" in its original digital format, before ever passing throughan D/A converter, and is not actually taken through the sound card? Perhaps there was some confusion or an ambiguity taken the wrong way? That would eliminate the other person's argument here, if I'm not mistaken. I'm no expert here, I'm just trying to comprehend how this analog hole stuff works.

      --
      This is a hacked account, for which the owner can not be held responsible.
    8. Re:Sound quality by drgonzo59 · · Score: 1

      Once the sound is digitized, a software package can be used to edit and manipulate it. The only thing that should not happen inside a typical PC case as any A/D or D/A conversion. Only digital streams should be manipulated.

    9. Re:Sound quality by Hal_Porter · · Score: 1

      Audiophiles don't rip CDs. They buy the LP gold masters and a pressing plant.

      --
      echo -e 'global _start\n _start:\n mov eax, 2\n int 80h\n jmp _start' > a.asm; nasm a.asm -f elf; ld a.o -o a;
    10. Re:Sound quality by Metteyya · · Score: 1

      So the signal is "intercepted" in its original digital format, before ever passing throughan D/A converter, and is not actually taken through the sound card?

      Exactly :) The name AnalogWhole may seem confusing, because I think they wanted to show analogy to "plugging" everything in and out.

    11. Re:Sound quality by Al+Dimond · · Score: 1

      He's right and it is applicable here. I R'd TFA and it said that the Windows mixer component is used to instruct the sound card to loop output back to input. I did that once on my computer as an attempt to get a sample of a MIDI file. You don't have to be an audiophile (I'm not) to hear that the quality of playback is much worse than the original playback of the MIDI file. It would be exactly the same if the program manipulated audio streams like JACK. You can, in fact, hear noise in the recordings corresponding to activity on other parts of the motherboard (I have an onboard sound card, and a poor one at that).

      Linux apps that can use JACK need to be written to support JACK's API. I think JACK has a kernel component as well, but I may be wrong. At any rate, the most straightforward way I can think of for a program to simply store a *digital* stream from any program directly to a file would be to write a "fake" audio driver that writes to disk. For programs with output plugin support (Winamp and similar programs come to mind) you could create a "disk writer" plugin (and this has been done), but in the general case you'd have to write a general audio driver (I'm pretty sure that's been done as well).

      If some level of DRM decoding is done at the OS (including kernel or libraries) or hardware level, and if the OS does not allow unsigned drivers, this approach is thwarted.

    12. Re:Sound quality by magetoo · · Score: 1
      It's just that signal - still in digital form - is received by another app, that's all.
      http://analogwhole.com/?page_id=8
      Windows Media Player does the tough job of converting the 1s and 0s particular to that codec the music was stored as into an analog output that is played through the sound card. While the song is playing, AnalogWhole re-routes this analog signal back into the recording input of the sound card. As it is recording the music it stores it as an MP3 file.
  2. Still loss of quality by amplusquem · · Score: 5, Informative

    It is still looped through the sound card, so while quality may still be "excellent", there is still loss. I would rather use a program such as QTFairUse which doesn't lose any sound quality.

    1. Re:Still loss of quality by omeomi · · Score: 3, Insightful

      It is still looped through the sound card, so while quality may still be "excellent", there is still loss.

      There's also loss do to re-compressing an already compressed file as an MP3. Overall, it's not the best of option...especially given the horrid quality of most consumer-level ADC's.

    2. Re:Still loss of quality by westyvw · · Score: 1

      If your starting with crap like WMA then moving it to lesser but still crap MP3 why would you care?

    3. Re:Still loss of quality by Threni · · Score: 5, Funny

      > It is still looped through the sound card, so while quality may still be "excellent", there is still loss.

      mmm...but just listen to that lovely analog warmth! I'll take that over digital accuracy anyday...

    4. Re:Still loss of quality by gameforge · · Score: 5, Interesting

      I know a number of audiophiles who detest MP3s. I've tricked them into saying that the actual CD was an MP3 and the MP3 I ripped from that CD was the real CD. They couldn't actually tell a difference and were taking guesses.

      If you use LAME, set your Q to 9. A 320kbps MP3 with Q=1 and 320kbps mp3 with Q=9 are WILDLY different, while both the same bitrate and same size. Whatever garbage MP3 files you have, re-encoding them as 320kbps/Q9 files isn't going to make them sound any worse to 99.9% of humans. Of course it takes more time to encode them this way.

      Another point, not for you, but for some of your parent posts - think about a soundcard with a digital out. That means, the bits get decoded and sent to the amp - if the amp (or whatever you plug the digital line into) can capture the bits, you've got a perfect/lossless rip - no DAC was involved. Volume controls and DSP's may change the bits somehow, and it will take playing-with to get it right... but it will produce satisfactory results once you do.

      I would test this for people, but I own (and will always own) absolutely ZERO DRM content.

      I own a Creative SoundBlaster Audigy... I know even a cheap SBLive! can do this... I would try the following to get a pure digital copy, in this order:

      1. Play a DRM'd file, set the recording channel to "What U Hear", and record. If that doesn't work...
      2. Get a LiveDrive (plugs into SB Live's & Audigy's) cheap on eBay, and an optical cable... then plug optical out into optical in and try to record the optical in. If that doesn't work...
      3. Get two computers, one with a digital out and one with a digital in. Try it that way. If that doesn't work...
      4. Uninstall iTunes or whichever thing is giving you this unplayable worthless crap to begin with, and tell their distributor to go to hell. Then take your stereo equipment and hurl it at Sony-Poo's nuts, and sing to yourself until a better solution comes along.

      I can actually guarantee positive results with that last one.

    5. Re:Still loss of quality by krunk4ever · · Score: 1

      The current version of QTFairUse (2.4) doesn't work with the newest iTunes, however there's currently 2.5 beta 1 which is awesome. I don't know if these features were also in 2.4 since I jumped from 2.3 to 2.5 beta 1 (because 2.4 didn't work after the iTunes upgrade), but 2.5 now includes the ability to not just strip the DRM from the m4p files and redo the ID3 tags, it even has the option of backing up the the files, removing the DRM version from your library and adding the new DRM-free version back into your library, transfer your rating from the old file to the new file, and even finding all the playlists that had that song and re-adding the song back to these playlists.

    6. Re:Still loss of quality by krunk4ever · · Score: 1

      I actually forgot to mention that the best new feature is the ability to recreate the file without playing the file at 1x speed. In previous versions, it had to play the entire song through before it was done recreating the new m4a file. However with this new version, it was able to recreate the DRM-free version in just a few seconds. I guess they found a way to get iTunes to play the file really fast because it still needs iTunes to play the file, but no audio comes out.

    7. Re:Still loss of quality by malsdavis · · Score: 1

      What sort of level of loss are we talking about here?

      Is it actually going to reduce the quality below that of around say 192k?

      Personally, for anything higher than that I really can't tell the difference anyway.

    8. Re:Still loss of quality by Anonymous Coward · · Score: 0
      I know a number of audiophiles who detest MP3s. I've tricked them into saying that the actual CD was an MP3 and the MP3 I ripped from that CD was the real CD. They couldn't actually tell a difference and were taking guesses.

      A percentage of audiophiles are utter morons, deaf as they are clueless.

      Still your test largely depends on source material and listening conditions. Try it with a clasical piece in a good listening room at medium volume.

    9. Re:Still loss of quality by chazwurth · · Score: 1

      I know a number of audiophiles who detest MP3s. I've tricked them into saying that the actual CD was an MP3 and the MP3 I ripped from that CD was the real CD. They couldn't actually tell a difference and were taking guesses.

      Were you playing the content on a crappy (read: normal, average, home) sound system? In my experience the difference in quality between mp3 and CD audio is extremely clear if you're listening on a hi-fi system. And I'm not an audiophile.

      --
      The plural of 'anecdote' is not 'data'. --Dan Kaminsky
    10. Re:Still loss of quality by gameforge · · Score: 1
      A percentage of audiophiles are utter morons, deaf as they are clueless.
      No, I checked - they're not those kind, it's okay.

      I'm sure that still isn't good enough for you, is it.

      I agree with you about the classical music, the utterly most pleasing sound I've ever heard was a performance of the Colorado Symphony Orchestra this Spring; it featured world reknown (and GORGEOUS) cellist Wendy Werner. She's the Paganini of the cello. I have both CDs and MP3's of classical music; I also have a 5.1 studio on my computer. I stand by my observations.

      If you really don't believe me, try it yourself. I'm not here trying to offer empirical evidence... I'm offering my experience. If you have different experience, let's have it already!
    11. Re:Still loss of quality by westlake · · Score: 1
      I know a number of audiophiles who detest MP3s. I've tricked them into saying that the actual CD was an MP3 and the MP3 I ripped from that CD was the real CD. They couldn't actually tell a difference and were taking guesses.

      This doesn't tell me much.

      Edison used "blind" tone tests with live singers and musicians to demonstrate the quality of his acoustic recordings and phonograph players. But he was careful to chose just the right solo voices and instruments.

    12. Re:Still loss of quality by CryoPenguin · · Score: 1

      If you use LAME, set your Q to 9. A 320kbps MP3 with Q=1 and 320kbps mp3 with Q=9 are WILDLY different, while both the same bitrate and same size.

      Huh? Lame's manpage says that -q 0 is the slowest and highest quality, while -q 9 is the fastest and lowest quality. Do the win32 frontends remap that range or something?

    13. Re:Still loss of quality by ben+there... · · Score: 1
      I would rather use a program such as QTFairUse which doesn't lose any sound quality.

      And I'd rather use FairUse4WM than QTFairUse. It is much faster because it's a standalone decrypter that doesn't rely on iTunes API or hooking into the iTunes process. At least 4x faster, subjectively and IIRC. It also doesn't require a reencode because it's just removing the DRM.

      I'd guess that the only use for AnalogWhole is for files that for some reason don't work with FairUse4WM.
    14. Re:Still loss of quality by gameforge · · Score: 1

      Oops, I got them mixed up again.

      Yeah, it's Q=0. :-) Sorry.

    15. Re:Still loss of quality by p3d0 · · Score: 1

      I like the fact that someone modded this Insightful. Very funny.

      --
      Patrick Doyle
      I mod down every jackass who puts his moderation policy in his sig. Oh, wait a sec....
    16. Re:Still loss of quality by gameforge · · Score: 1

      The two I'm thinking of both used studio headphones. I've done comparisons with other people who weren't sound engineers just on my computer; I have a Carver M400 cube amp with two Advent towers on my front channels, and decent stuff (but ultimately junk) on my rears. The Advent towers are above average; but the Carver cube is far superior to stuff you find a Best Buy. I would call it a lower end studio amp (not even a volume control on it; you plug it in and it's on, because it's supposed to plug into the power supply of a preamp. I use my SoundBlaster Audigy 2 for a preamp, so it can result in some obnoxious surprises depending on where the card's mixer volume is set). It's your typical guy-pad type setup that no wife would ever approve of or be caught dead with in her living room. :)

      One of the studio engineers actually proved that to my ear anyway, LP's on great record players with expensive stylus' & mics actually sound superior to CDs. I'm a believer, too.

      If you really want to know (getting tired of saying this) try it for yourself; LAME is free. I wasn't offering scientific research, just my experience.

    17. Re:Still loss of quality by Anonymous Coward · · Score: 0

      A couple of years ago, I archived all of my CDs to my hard drive and wanted to choose the best codec. I did a little comparison between musepack, LAME, and ogg vorbis, and the results were surprising. I ripped a Nirvana song to wave format and then encoded it at ~200kbps variable bit rate in all three formats. I then queued up the wave file and all three encoded files in Winamp and did a listen test by switching back a forth between the wave file and the encoded files.

      I used a Nirvana song because their songs have a very wide frequency range and typically produce lots of artifacts at low (~128) bit rates.

      Compared to the wave file the LAME file's low-end frequecies were higher.
      The musepack file's mid-level frequencies were higher.
      The ogg vorbis file sounded exactly like the wave file.

      While none of the codecs produced any audible artifacts, the differences in frequency response lead me to choose ogg, as it did the best job at reproducing the original sound.

      I also agree about sound cards DA converters being decent enough. I've done transfers of my vinyl records using my cheapy SBLive Value card and the results were just fine.

    18. Re:Still loss of quality by tlhIngan · · Score: 1

      LAME has a nice set of settings called "presets" that have all the best-quality settings put in them (this was done using suggestions based on r3mix evaluations). There are several of them. Just use --alt-preset or --preset, with "standard", "extreme" or "insane" (in increasing order of quality). This enables VBR, which keeps the files smaller (no need for 320kbps when you don't need it, but gives LAME the flexibility to go to 320kbps). I use Extreme, and it tends to average between 192 to 256kbps. I can't tell the difference, but 128kbps is painful.

      I used to use tons of command line options, but now, I just use --alt-preset extreme. Works great.

    19. Re:Still loss of quality by westyvw · · Score: 1

      My point was that taking a lossy format and converting it into mp3 with make that mp3 bad. Lossy --> lossy is MUCH worse then CD --> MP3.

      However I can, and have proven to others, that I can tell when I hear a WMV VS a MP3.

      Telling the difference between a high bitrate MP3 and source is much harder, but MP3's usually sound "harsher" and somewhat empty compared to the source.

    20. Re:Still loss of quality by Anonymous Coward · · Score: 0

      congratulations, you chose the format supported by ~5% of portable music devices and 1% of car digital audio devices! have fun listening to all that music while tied to your computer though

    21. Re:Still loss of quality by Anonymous Coward · · Score: 0
      I own a Creative SoundBlaster Audigy... I know even a cheap SBLive! can do this... I would try the following to get a pure digital copy, in this order:

      Nope - none of these cards are bit-perfect digital. You can test them out for yourself - they drop bits frequently. Try capturing the output on a good digital recorder (even the cheap jb3 has been shown to be bitperfect, so try that), and then transferring it back to your computer and comparing it to the original audio file. You will find they are different with the cards you listed. You need something like an audiophile 24/96 for bitperfect digital.

    22. Re:Still loss of quality by gameforge · · Score: 1

      Maybe, but I'm hesitant to believe that. I'm a digital circuits guy... individual bits don't just disappear. They either flip or get corrected, or masses and droves of them disappear. It seems weird that their product would randomly and "frequently" flip bits. Please post an article or something outlining more clearly exactly what you're talking about.

      Why don't I hear clicking & popping when I play my guitar (got a digital effects processor) or use my PS2? Both connect optically to my Audigy 2 Live Drive... seems you'd hear it if it was "frequently" losing data, although I'm not a hardcore sound guy (and am quite happy not being one) and wouldn't necesarily know what to listen for.

      This card has an Emu DSP on it. That same DSP is used on other cards which are very similar to the SoundBlaster but are geared for studio use and come with far superior drivers. I'd use those drivers (supposedly there's a way to make it work) but I'd lose EAX and 3d sound... anyways, the DSP does all the work. Seems unlikely that it doesn't work.

      If you're talking about the ADC, then while that wasn't at all what I was talking about, I couldn't agree with you more. That's why the one on this card is 24-bit 192KHz; you'll end up with 96K samples per second of what you want.

      If you're talking about the DAC, I'm still suspicious, as it would directly affect what you hear.

      If you're talking about this card on a slow computer, the errors are the software dealing with bus bandwidth, not the card.

      I'm not purchasing any equipment (in fact I'm not even going to get out of my chair) to prove anything. I've used this chip in some form for four years and have been extremely pleased with it.

      So anyway, sorry for being longwinded... please cite a source which explains this phenomenon pertaining to SoundBlaster Live and Audigy products. In the meantime, trust me, this card will work just fine for ripping low quality DRM crap, and I stand by everything I originally said. Just equate "pure digital copy" to "your pet Dolphin couldn't even tell the difference".

    23. Re:Still loss of quality by evilviper · · Score: 1
      Whatever garbage MP3 files you have, re-encoding them as 320kbps/Q9 files isn't going to make them sound any worse to 99.9% of humans.

      Actually, you're wrong there. Certainly encoding from lossless copies to MP3s at highest bitrate and -q0 (NOT Q9!) will sound perfect to most everybody.

      HOWEVER, that is certainly NOT the case when repeatedly reencoding.

      Use any of the best lossy audio codecs in the world, and encode with the highest possible quality, then decode and reencode the file 10 times... IT WILL SOUND LIKE COMPLETELY CRAP, no matter what.

      The situation gets worse with every generation. However, even with only 2 rounds of encoding/decoding, the number of people who will hear distortions is much, much higher than 0.1%. I'd just guess it would be along the lines of 5% or so.

      think about a soundcard with a digital out. That means, the bits get decoded and sent to the amp - if the amp (or whatever you plug the digital line into) can capture the bits, you've got a perfect/lossless rip

      Unlikely. CDs are 44.1KHz, while soundcards usually require 48KHz, meaning at the very least, the audio is being resampled, which isn't a lossless process.

      Volume controls and DSP's may change the bits somehow, and it will take playing-with to get it right... but it will produce satisfactory results once you do.

      "Satisfactory" of course, but nowhere near perfect/lossless.

      I would test this for people, but I own (and will always own) absolutely ZERO DRM content.

      Files without DRM sound the same on the digital output as files without. This is a nonsense excuse.

      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    24. Re:Still loss of quality by Anonymous Coward · · Score: 0

      If you resample 44.1 to 48, you get exactly what you started with... 44.1 khz quality at 48 khz. Your ears cannot hear a difference between the two, because there *technically is no difference*. Your soundcard just samples it 3900 times/sec faster, and extra "nothing" is inserted into the PCM bitstream.

    25. Re:Still loss of quality by KDR_11k · · Score: 1

      That's why my PDA is my "mp3" player, just install a different player app and it supports ogg.

      --
      Justice is the sheep getting arrested while an impartial judge declares the vote void.
    26. Re:Still loss of quality by gameforge · · Score: 1
      Your post is complete flamebait, but it's the wee hours of Sunday morning, so I'll bite. Feel honored; it makes you a masterb.. well, anyway. First off, I'll give you the Q0 thing. Someone corrected me earlier. Blame lame; their numbers are backwards.

      Actually, you're wrong there...even with only 2 rounds of encoding/decoding, the number of people who will hear distortions is much, much higher than 0.1%. I'd just guess it would be along the lines of 5% or so.

      Okay 95% of everyone won't claim that their garbage MP3 sounds any worse when reencoded on the highest parameters. I wasn't being empirical, but I didn't realize that wasn't obvious...

      Of course, I'm sure you're aware that switching lossy formats every time is going to slow this iterative decay down. Think of a lowish quality WMA - export it to a wave & listen to it, take it for what it is. You said yourself a 320/Q0 MP3 from a lossless source is going to be difficult to tell apart. Now consider your output WAV as a lossless source, and think of all the other 320/Q0 MP3's you've witnessed: you're breaking into that top 5% man! That high quality 320/Q0 MP3, like all the others you've seen, will sound no different. Likely, a 320/Q9 MP3 would sound no different.

      Since MP3 is not a DRM format and this article doesn't apply to reencoding WMA's as WMA's or AAC's as AAC's, I stand by my original estimate of 99.9%. Either way, even according to your research, 19 out of 20 would agree. Does my point not stand?

      soundcards usually require 48KHz, meaning at the very least, the audio is being resampled, which isn't a lossless process.

      Upsampling 16/44.1 to 16/48 is a lossy process. Why would I care, I have a 24-bit/96KHz soundcard, and SPDIF has no predefined sample rate. Incidentally, I can record from the ADC on this card at 192KHz.

      Low and behold, it records any damn thing I play into it just fine, and has for years.

      "Satisfactory" of course, but nowhere near perfect/lossless.

      Umm, no. VERY close. It's got a DSP so you have to turn the Bass & Treble to dead center, turn on 2 channel output instead of 5.1 or whatever, etc. You also have to have enough bus bandwidth to get it into your memory on time; if you don't run a bunch of garbage in the background, that's easy, my computer's dealt with it fine and it's four years old. I can play a MIDI track and play guitar into my pedalboard, which hooks up to the SoundBlaster with SPDIF, turn on reverbs & chorus' on the DSP, and use What-U-Hear to record the whole mess. I mean, I do this at least once a week. With SPDIF to SPDIF, you just have to make sure both input & output levels are maxed. Long story short, this means the data all gets multiplied by a whopping 1.0.

      Don't take my word for it. I know your $0.26 stock soundcard doesn't do that. Go buy a $10 SBLive and see for yourself.

      I'm losing my faith in you, Mr. evilviper.

      Files without DRM sound the same on the digital output as files without. This is a nonsense excuse.

      See, now clearly, you're not trying to convince me you're an expert. You're just trying to get me to flame you!

      Clear your head a little, slow down, don't take on the "I'm gonna rip your post to shreds whether I'm wrong or not" mindset, and think about what I must have meant for a second.

      If I play an MP3 and record it with What U Hear, I'm going to get a duplicate of it. But Microsoft or Creative's little DRM smart code might kick in and prevent me from doing this when playing a protected AAC file or WMA file. Similarly, if I play a WMA file and hook up SPDIF-out to mine or another computer's SPDIF-in, the SPDIF "don't copy" flag might be set, and that may not work either. SPDIF does have a "don't copy" flag.

      Since I don't own any DRM material, I have no way of testing this do I? I know perfectly freaking well that I can record unDRM'd shit t

    27. Re:Still loss of quality by antispam_ben · · Score: 1

      Is it actually going to reduce the quality below that of around say 192k?

      It's rather hard to compare DAC/ADC distortions and analog noise with MP3 encoding/decoding artifacts, but 192k is significantly better than the commonly used 128k rate, and (even with my tin ears) I can hear the difference between those two. But that means you can hear cheap soundcard noise and distortion BETTER at 192k. I think the 128k listeners wouldn't be bothered by the difference between a built-in soundcard and a $100 "semi-pro" studio soundcard.

      As a generalization, "consumer" (whether no-name or Creative/Soundblaster/SBLive) soundcards suck. A few years ago I tried using several I had lying around for recording LP's (RECORDS on a TURNTABLE!), and there were buzzes and noises I could hear through the LP's background noise. I finally spent $150 (the price at the time) for a Delta Audiophile 2496 (it's not that it's more bits, I still transfer LP's at 16/44, it's that the whole thing is designed to reduce digital trash injection into the analog signal, uses better op-amps, and in general with better quality audio in mind), and haven't looked back.

      --
      Tag lost or not installed.
    28. Re:Still loss of quality by antispam_ben · · Score: 1

      Maybe, but I'm hesitant to believe that. I'm a digital circuits guy... individual bits don't just disappear. They either flip or get corrected, or masses and droves of them disappear. It seems weird that their product would randomly and "frequently" flip bits. Please post an article or something outlining more clearly exactly what you're talking about.

      It's not that it "flips" random bits, is that it does sample rate conversion (so that it does all its internal operations at 48k samples per second), which cannot be done without "changing bits." The SBLive and Audigy cards to ALL their internal processing at 48kHz. A digital output of a 44.1k signal through one of these cards gets converted to 48k and back to 44.1k. The bits don't come back the same. Whether the SOUND is similar enough to the original is a different argument.

      So anyway, sorry for being longwinded... please cite a source which explains this phenomenon pertaining to SoundBlaster Live and Audigy products.

      http://pages.sbcglobal.net/hamakerd/sbsrc/

      --
      Tag lost or not installed.
    29. Re:Still loss of quality by kg261 · · Score: 1

      There is some loss of quality, but one idea is to merge two or more files created from analog to create a better version. I have tried this be recording from a CD two times, then writing a program to align the waveforms as much as possible. The comparison part worked: I could see that either the sound card crystal or the CD crystal has drifted and the error between the waveform increases as time goes on as well as the amplifier noises. The next step is to continuously match the waveforms to factor out variations in the various sampling rates.

    30. Re:Still loss of quality by evilviper · · Score: 1
      Your ears cannot hear a difference between the two, because there *technically is no difference*.

      Ridiculous. 44.1 doesn't go into 48 evenly, and that's all there is to it. You can't just insert samples of "nothing" in a digital audio signal and get an audio stream.
      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    31. Re:Still loss of quality by evilviper · · Score: 1
      Your post is complete flamebait,

      No, it's the facts. You just happen to be a flamer, so anything you don't like is flaimbait to you.

      I, however, am going to avoid all of your flames and rantings.

      Of course, I'm sure you're aware that switching lossy formats every time is going to slow this iterative decay down.

      It will TRADE that delay for OTHER artifacts. Typically, discarding MUCH more of the audio, and now having the artifacts of both audio codecs.

      Now consider your output WAV as a lossless source,

      It isn't. It has lost most of the waveform. That's the whole reason why you don't call lossy codecs "lossless".

      That high quality 320/Q0 MP3, like all the others you've seen, will sound no different.

      No, it won't. Not at all. That you personally can't hear the difference can't possibly change that fact.

      SPDIF has no predefined sample rate.


      "SPDIF interface supports 3 standard sample rates: 48kHz, 44.1kHz and 32kHz. All other sample rates are impossible to transmit. Nevertheless, most audio cards support only 48kHz output."
      http://ac3filter.net/forum/viewtopic.php?t=10


      "Allowed sampling frequencies (Fs) of the audio:
              * 44.1kHz from CD
              * 48 kHz from DAT
              * 32 kHz from DSR "
      http://www.epanorama.net/documents/audio/spdif.htm l


      Good enough?

      Don't take my word for it. I know your $0.26 stock soundcard doesn't do that. Go buy a $10 SBLive and see for yourself.

      I've been doing digital audio back when you were probably still in diapers. It's just a shame you don't know how stupid most all of your claims sound to anyone who knows anything about digital audio, PC soundcards, lossy audio encoding, etc.

      the SPDIF "don't copy" flag might be set, and that may not work either. SPDIF does have a "don't copy" flag.

      SCMS is entirely ignored by every soundcard I've ever come across (you've clearly never tried), so you'll have to try harder.

      I know perfectly freaking well that I can record unDRM'd shit this way, why would I test that?

      Well, for one thing, to prove that what your getting isn't anywhere near lossless, even though you think so.

      I don't know what little goofball stuff happens in this process....

      There's plenty you don't know, yet you feel the need to give advice on these very subjects.

      but like I said to him, your pet Dolphin isn't going to know the difference,

      This is just sad at this point. You're convinced of a lot of things you've clearly never checked on. What's your evidence for this statement, and don't even try to tell me it's your years of experience.
      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    32. Re:Still loss of quality by smcdow · · Score: 1
      A 320kbps MP3 with Q=1 and 320kbps mp3 with Q=9 are WILDLY different...

      And "lame '--vbr-new -h --preset standard" will save a butt-load of disk (or iPod) space, and will sound great for about 99% of the non-audiophile-nerdboys. Not only that, but it's is how all the MP3s available from Emusic.com are encoded (non-DRM, too!). Highly recommended.

      --
      In the course of every project, it will become necessary to shoot the scientists and begin production.
    33. Re:Still loss of quality by Anonymous Coward · · Score: 0

      Nice attempt at a troll.

      I've never owned a portable music player, nor do I have an plans to own one in the near future, but If I even did own one, it would be one that does support ogg vorbis (and FLAC too), like the Neuros audio player, which btw can transmit music to your car stereo via FM, so your point is...rather pointless.

    34. Re:Still loss of quality by gameforge · · Score: 1

      It will TRADE that delay for OTHER artifacts. Typically, discarding MUCH more of the audio, and now having the artifacts of both audio codecs.

      De c ay Viper, decay... before you master flamebaiting, perhaps master reading comprehension first? Incidentally, I just tried it. I played a not-so-perfect OGG file I have of Comfortably Numb/Pink Floyd into a WAV and encoded it as an MP3 with LAME. It sounds as good as the OGG does. Anyone can try this for themselves; you don't have to take my word for it.

      I've been doing digital audio back when you were probably still in diapers. It's just a shame you don't know how stupid most all of your claims sound to anyone who knows anything about digital audio, PC soundcards, lossy audio encoding, etc.

      Uh huh (flamebait). First, digital audio wasn't at the consumer level when I was in diapers. My first sound card was an 8 bit original SoundBlaster; then I got a Pro Audio Spectrum 16, an SB AWE32, AWE64, SBLive, and SBAudigy. I also had a Diamond card for a while and built a SoundBlaster clone with a buddy. Still own all of them except the Diamond card, and they all still work.

      Now, for once. Tell me about your first experiences with digital audio. What was your first card? Did you ever write a DOS driver for SoundBlaster compatible cards? I did; I can send it to you, if you want. I wrote one for a game before Sean Hargreaves wrote Allegro. I realize this is irrelevant to the discussion, but it should demonstrate how long I've been doing this. If you want me (or anyone) to believe you're so much more experienced, you're going to need to cough up some real life examples and anecdotes.

      And I don't even claim to be an expert. If I did, quote me.

      Second, you clearly don't work with SPDIF much, anyone with a multimedia PC has done these things plenty of times before. For instance, me; I'm just a user. I would have no way of knowing any of this if I hadn't tried it myself. I do this stuff all the time because I'm a musician. Give me your e-mail, I'll send you some of my work. I can even show you a video of this very card recording stuff at amazing quality, if you want. If you're that skeptical. But since you're a troll, I'm sure you'll decline. Are you a musician? Do you record stuff at least once a week on your 48KHz hoopty card? I'd love to hear the things you record...

      If you'd quit being a troll, I'd even be willing to send you my old SoundBlaster Live for the cost of shipping plus $15 and you can see for yourself. Comes with a LiveDrive and everything; we can go through whatever payment service you feel comfortable with.

      It isn't. It has lost most of the waveform.

      (sigh) Reading comprehension thing again. Re-read what I wrote please.

      Good enough?

      Good enough for obsolesence. Go head over to Creative Labs and tell me how they put 24/96 SPDIF on their card? Better yet, buy one (like the one I offered you) and see for yourself. But go ahead & splain this, expert:

      S/PDIF is used to transmit digital signals of a number of formats, the most common being the 48 kHz sample rate format used in DAT, and the 44.1 kHz format used in CD audio. In order to support both systems, as well as others that might be needed, the format has no defined data rate. Instead the data is sent using Biphase Mark Code, which has either one or two transitions for every bit, allowing the original word clock to be extracted from the signal itself.

      Think of it this way: either SPDIF doesn't have a predefined sample rate, or Creative's SPDIF goes way above the standard. Meaning either way, you're incorrect. Do some math: your source even says the bandwidth ranges up to 6MHz; that's enough to do 24/bit 5.1 at about 44KHz, or stereo 24-bit at over 200KHz.

      SCMS is entirely ig

    35. Re:Still loss of quality by shaka · · Score: 1

      If you use LAME, set your Q to 9. A 320kbps MP3 with Q=1 and 320kbps mp3 with Q=9 are WILDLY different, while both the same bitrate and same size. Whatever garbage MP3 files you have, re-encoding them as 320kbps/Q9 files isn't going to make them sound any worse to 99.9% of humans. Of course it takes more time to encode them this way.


      I'm not saying you are necessarily wrong, but according to the documentation for my lame (version 3.96.1):
      -q qual
                    0 <= qual <= 9
       
                    -q 0:
                    use slowest & best possible version of all algorithms. -q 0 and
                    -q 1 are slow and may not produce significantly higher quality.
       
                    -q 2:
                    recommended. Same as -h.
       
                    -q 5:
                    default value. Good speed, reasonable quality.
       
                    -q 7:
                    same as -f. Very fast, ok quality. Psycho acoustics are used
                    for pre-echo & M/S, but no noise shaping is done.
       
                    -q 9:
                    disables almost all algorithms including psy-model. Poor qual&#8208;
                    ity.
      --
      :wq!
    36. Re:Still loss of quality by gameforge · · Score: 1

      Yes, that was pointed out to me twice already.

      I always get the numbers backwards... it is -Q 0. You'd think the highest quality would be -Q 9, but no...

    37. Re:Still loss of quality by gameforge · · Score: 1
      I appreciate your response, thank you for giving me some info.

      From your source, regarding digital loopback @ 48KHz:

      This test should test everything in the loopback path except the converters. The result is actually fairly interesting. All the measures are "excellent," except there is an unexpected wiggle in the frequency response which RMAA 5.2 considers "average." This might raise some concern if one is using the S/PDIF output of the Live! (this result may or may not apply to the Audigy-series cards, whereas all previous results probably do) for serious purposes.
      So, you may want to upsample to 24/96 yourself first.

      I do agree that resampling changes bits; however, going from 16/44.1 to 24/96 and back isn't going to compromise the sound you hear if you do it only once. According to Creative, their cards upsample everything to 24-bit internally; they claim this makes your MP3's sound better, among other things.

      Also, your source mostly discusses the analog functions of the Emu10k1 cards; although I'm surprised to learn that their SPDIF lines aren't as perfect as Creative advertises.

      I still contend that average DRM ripping humans wouldn't know the difference, even a well trained ear with expensive studio headphones.

      Thanks again for the info. If you have any more you'd care to post, I'm inclined to know more.
    38. Re:Still loss of quality by magetoo · · Score: 1
      Ridiculous. 44.1 doesn't go into 48 evenly, and that's all there is to it.
      Well, I disagree. Parent mentioned that "your ears cannot hear a difference"; don't forget that our ears and brains throw away lots of information. As long the errors are outside our range of hearing, or of a kind we can't percieve (phase...) there is, for musical purposes, no difference.


      You're right that you won't end up with a "pure digital copy" (like GGP claimed), and you are right that this sort of thing should be avoided if possible, of course.

      My point is that it's not that simple, and while there might be distortion, it's not necessarily audible.

      You can't just insert samples of "nothing" in a digital audio signal and get an audio stream.
      Obviously. You need a low-pass filter too. (This is actually a common method of resampling with interpolation, in case anyone was wondering...)
    39. Re:Still loss of quality by evilviper · · Score: 1
      Well, I disagree. Parent mentioned that "your ears cannot hear a difference";

      Well you seem to be disagreeing with something I haven't yet said...

      Whether you can hear it or not is besides the point. In fact, few people will be able to hear the difference between the analog and digital version anyhow. I have just trying to make the point that digital output isn't necessarily lossless as so many people seem to believe, and the benefits of digital instead of analog in this case are probably nil.

      As long the errors are outside our range of hearing, or of a kind we can't percieve (phase...) there is, for musical purposes, no difference.

      Resampling applies to the full waveform, so it's not just affeting sub or super sonic sounds. Whether you can hear it or not depends on how good your ears are. It's certainly possible for someone with good ears to hear it.

      Obviously.

      Did you actually read the post I was replying to?
      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    40. Re:Still loss of quality by evilviper · · Score: 1
      De c ay Viper, decay...

      A trivial and obvious typo. But troll away.

      I played a not-so-perfect OGG file I have of Comfortably Numb/Pink Floyd into a WAV and encoded it as an MP3 with LAME. It sounds as good as the OGG does.

      Your ears (or perhaps audio equipment) suck. Not everyone else's does.

      First, digital audio wasn't at the consumer level when I was in diapers.

      And? I've been doing PROFESSIONAL studio work for a very long time now.

      Tell me about your first experiences with digital audio. What was your first card?

      Haha. Digital audio predates personal computers by several decades, thanks.

      Second, you clearly don't work with SPDIF much, anyone with a multimedia PC has done these things plenty of times before.

      Recording audio via SPDIF is trivially EASY. Getting a bit-perfect copy is quite a bit more difficult with PCs. You clearly fall into the first category, and are convinced anything you can't hear, can't possibly exist or matter, and that's what you keep arguing.

      I'd even be willing to send you my old SoundBlaster Live for the cost of shipping plus $15 and you can see for yourself.

      I've got plenty of them lying around here, and anyone who thinks Creative cards are the best available hasn't got any idea what they are talking about.

      "has no defined data rate."

      DATA RATE IS NOT SAMPLE RATE. GET IT THROUGH YOUR HEAD.

      One is the samplerate of the waveform, the other is the number of bits per second. That means you can send compressed audio (AC3/DTS) just as easily as uncompressed PCM over SPDIF. But guess what, no matter the bitrate, AC3 and DTS are both ALWAYS 48KHz.

      Really? Which cards? SCMS comes from DAT.

      I have no idea what you're trying to say here, but I doubt you do as well. SCMS has nothing to do with DAT, other than the fact that DATs happened to commonly have SPDIF connections. Minidiscs, CD recorders, etc., all use SCMS. That is the SPDIF "security" you think you know about. There is no other.

      But hey, you're the expert. What do I know. I'm more than done now. Have the last word if it will make you feel better.

      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    41. Re:Still loss of quality by gameforge · · Score: 1
      First, I'll kindly ask you to leave your haste and arrogance elsewhere the next time you choose to start a mostly fallacious argument. It only makes you more difficult to argue with; I can't tell if you misread what I wrote, or re-wrote it to be a straw man. You've (quite creatively) misread every one of my posts. You may be A) Very hasty and yet wish to argue with anything that moves, B) very stupid, and/or C) you're just trying to aggrevate me and get me to flame you (since I'm trying not to go ad hominem too much here, I'll go with A & C).

      Glad to see you're dodging all the demands for real info. Since you'd like me to have the last word, I'll summarize everything I can gather from your nearly identical replies:
      • You have no personal experience to talk about (nyeh, I'm a studio professional, nyeh, doesn't count - what studio you work for/in/with? What kind of console did they have? The one nobody's heard of, right? Plugging in your DVD player is not professional studio work)
      • I said you clearly had nothing to do with digital audio before PC's - I never said Creative Labs invented it, nor did I say it didn't exist before PC's. I think that was your 40-something'th misquote (which is making me suspicious).
      • You are a wise old man who used to work on UNIVAC's, so clearly, my hearing is much better than yours. My audio equipment came from a studio (go look up what a Carver M400 cube is); it's far more than adequate for what I do, complete overkill for low quality audio like the OGG file I described. Still lower end for the HiFi category, though.
      • You can't figure out that maximum data rate constitutes maximum sample frequency, nor can you read specifications clearly. "6MHz of bandwidth" (your source) refers to bits per second. Digital samples per second DIRECTLY converts into bits per second. If you really are a studio monkey (sorry "professional" studio monkey), then you have a very miserable understanding of digital circuits to say the least. Also, playing a sound out of a digital output on a PC doesn't send encoded AC3 (or DTS); most PC/SPDIF cards require extra equipment to do that (of course, you knew this already).
      • After several requests you won't tell me about these apparently limited/terrible/spectacular cards you use, which is why I contend that the Creative cards are so much better. They're mid level consumer cards; I know that.
      • You seem to know something about SCMS. But I'll gladly admit I don't know the first thing about it - only what I saw on Wikipedia. Once again, you don't have to be an angry moron to post stuff about it... whatever their "security" is, that's what I was referring to when I said I couldn't test playing DRM'd stuff into another SPDIF port. Second time I've written that...
      • Things you can't hear or detect sure seem to bother you a lot. What'd I say about your pet Dolphin?
      • You actually have no counterpoints to my original post. You keep offering all of these assertions about what does what, but your neglecting that I was only offering my experience and suggestions; I keep looking all this shit up on the fly to find out why it works for me when you claim it's not possible. You're the apparent expert, with your "professional" studio work... unfortunately, you're constantly ready to shove words in my mouth/post and then argue with me like I somehow posted inaccurate information.
      • FYI, entire sentences are not typos. You weren't paying attention, is what was going on there. Seems to be a thing with you.
      • And once again (though most likely not for the last time), I don't claim to be an expert.

      I'm sorry you've had enough. I know it sucks trying to dodge the bullet (i.e. the "hard" questions). Incidentally, as long as you keep feeding me drivel, I'll gladly keep spitting it back at you. I'm a pool guy in Denver and get a sweet check to do whatever I want in the Winter; it's not like I'm real busy for the next five months.
    42. Re:Still loss of quality by magetoo · · Score: 1
      I have just trying to make the point that digital output isn't necessarily lossless as so many people seem to believe, and the benefits of digital instead of analog in this case are probably nil.
      Not necessarily, no; and it doesn't magically solve all problems. But up- and downsampling can be lossless. I just don't see how you can so easily claim that there is a loss. (Unless you define "loss" as "anything that changes the bitstream". My definition is closer to "anything that removes (audible) information".)


      Yes, in reality obviously you will have to make a tradeoff; we don't have hardware with infinite floating point precision, etc, and so resampling will introduce some noise. But it's still not as easy as "that's all there is to it".

      As long the errors are outside our range of hearing, or of a kind we can't percieve (phase...) there is, for musical purposes, no difference.
      Resampling applies to the full waveform, so it's not just affeting sub or super sonic sounds. Whether you can hear it or not depends on how good your ears are. It's certainly possible for someone with good ears to hear it.
      Oh, it's affecting audible frequencies too, I'm not disputing that. But there's still no audible difference.

      Did you actually read the post I was replying to?
      I read the whole thread. I believe you were replying to the statement "Your ears cannot hear a difference between the two, because there technically is no difference".


      Anyway, we're probably not going to come to any agreement based on a Slashdot discussion, so I suggest we agree to disagree.

    43. Re:Still loss of quality by evilviper · · Score: 1
      I just don't see how you can so easily claim that there is a loss. (Unless you define "loss" as "anything that changes the bitstream".

      Anything that irreversibly changes it, yes (you can't resample and get exactly the same original bitstream). That is the DEFINITION of "lossless".

      Changes which can't be heard are known as "transparent".

      For the purpose of listening, transparency is fine (in which case analog is fine, too). But when you are talking about lossy encoding it afterwards, that will change how the output file sounds based on even the inaudible information in the bitstream.

      A good example of this is repeatedly reencoding a file with a high-bitrate lossy codec. After one round, you probably won't hear any difference, but with multiple rounds, you will. Even if you have terrible ears, after about 4 rounds of lossy re-encoding, it will sound terrible. Transparent lossy encoding is more destructive than resampling (though there are some pretty bad resampling algorithms out there) but it's can have similar effects that just aren't immediatly audible.

      I read the whole thread. I believe you were replying to the statement "Your ears cannot hear a difference between the two, because there technically is no difference".

      Yes, the latter part of that is certainly not true. There is a very real difference, despite the AC's claims to the opposite.
      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    44. Re:Still loss of quality by GWBasic · · Score: 1
      Another point, not for you, but for some of your parent posts - think about a soundcard with a digital out. That means, the bits get decoded and sent to the amp - if the amp (or whatever you plug the digital line into) can capture the bits, you've got a perfect/lossless rip - no DAC was involved.

      Actually, that's usually not true. In order to allow multiple programs to play back sounds simutaniously, sound cards internally convert everything to 48000khz before playback. They also allow for digital manipulation of the volume. Thus, all CDs, (and other formats that are ripped from CD) are upsampled from 44100khz to 48000khz before playback. They might also have their volume modified.

      The upsampling causes problems when playing back a DTS-CD on a computer because DTS-CDs require an unmodified digital connection to the reciever. The upsampling breaks DTS-CDs.

      Now, it is possible for a program to avoid this issue by directly controling the SPDIF. This is what happens with programs that can output Dolby Digital / DTS from a DVD or AVI. They bypass the soundcard's resampling (and volume control logic) to directly control the device. As a consequence, only one program can access the SPDIF at a time.

      Theoretically, it's possible to write a program that could play back non-upsampled 44100khz material. (Heck, you could even do it from a Winamp plugin.) At this time, I don't know of any programs that do it.

      Even if you can playback at non-upsampled 44100 through an SPDIF, there's no garuntee that resampling won't occur on a soundcard that records the SPDIF. Some of them upsample everything to 48000khz so they can mix all of the inputs; even if you're recording at 44100khz.

    45. Re:Still loss of quality by Anonymous Coward · · Score: 0

      oooooo! FM transmitter, arent you just the fake audiophile

    46. Re:Still loss of quality by Gr8Apes · · Score: 1

      Advent and Carver, not bad. :) I've tried a couple of things out as well, and this is on the systems in a couple of cars, as I play the original CD @ home after listening to 1 set of MP3s. 1st car - Pioneer deck from a while ago, cheesy cheap speakers - anything below 192 HQ is usually recognizable as MP3 if you knew the original CD version. Second car: much better sound system, MP3s below 256 are recognizable due to certain sound drop-offs and artifacts.

      Home stereo, Denon 3803, hooked up to a Denon CD/DVD player and some old cheesy Sony tower speakers (Never got around to upgrading them - wife nixed my old Pioneer towers, and my Dynaco's died - there's an old reference for you;) Most MP3s sound pretty awful.

      It really depends upon the music and waveform you're trying to compress. For instance, nice smooth waves like those from classical music actually seem to compress with less loss than heavily produced music, such as, say, Nine Inch Nails. (The Broken CD is broken in more ways than 1 if you try to compress it with MP3's codec.)

      As for LPs, I have what was considered a near the top record player (Pioneer LP45D I think, it's at home...I'm not) with a Shure V15 Type IV cartridge. So unless there's some magical LP player out there, a well-produced CD is far superior to a well produced LP. LPs always have an underlying crackle/hiss that detract from the "silent" parts of a musical piece. (The S/N ratio is about 50% higher for a CD than an LP - reference here and here)
      Oh, I also have a linear tracking arm for that record player. It makes no difference.

      To quell the obvious troll: there are those who prefer tube to transitor or digital amplifiers, and there are those who prefer vinyl to digital. It's a preference. As for 'x' sounds better on 'y' than 'z', anything produced badly for a particular format will sound worse than a better produced product on a lesser format. Some examples: Any of the Boston albums (initial pressing) CDs sound relatively terrible compared to their LP counterparts because the masters were originally produced for vinyl and accentuated the highs, to compensate for vinyl's softening of them. This was a problem with early CDs all around, and can still occassionally be seen in a badly produced CD. Other badly produced CDs are those, like some of Rush's latest CDs, that compress the dynamic range into the upper 10% to produce "loud" CDs, in hopes that that loudness will be noticed in airplay. Doing the same on vinyl will result in the same crappy sound.

      --
      The cesspool just got a check and balance.
  3. And the point is?? by Jane+Q.+Public · · Score: 2, Interesting

    Users have always been able to do this manually, if they had a decent recording program. Why the hoopla over a fancy software tool designed to do this one thing specifically? Does it save a few seconds? Further, this is really beside the point. DRM often still prevents users from making faithful digital copies of their own -- purchased, paid for, and legal -- media. This is a non-issue.

    1. Re:And the point is?? by Anonymous Coward · · Score: 0

      I guess the point is that now any idiot who can start wind'ohs and download iTunes for it can do this, whereas before one had to be slightly more computer-literate. Of course this is supposed to be "News For Nerds : Stuff That Matters" but nothing is as it seems as usual...

    2. Re:And the point is?? by Monsuco · · Score: 1

      Why are their not programs that do this for iTunes in the same way. They both cannot really be patched? An option of converting it from WMA protected to WMA unprotected would be good to minimize quality loss.

    3. Re:And the point is?? by magetoo · · Score: 1
      Why the hoopla over a fancy software tool designed to do this one thing specifically? Does it save a few seconds?
      Well, yes? And also (from TFSummary, no less):
      All meta data is transferred as well.
      That alone makes a hell of a difference.
  4. So... by Constantine+XVI · · Score: 2, Insightful

    So, it's just like using Audacity to record whatever goes through the sound card?

    --
    "I think an etch-a-sketch with an ethernet port would beat IE7 in web standards compliance."
    1. Re:So... by LordVader717 · · Score: 1

      As far as I know Audacity records the PCM stream, so there should be no loss of quality.

  5. thanks, but no thanks by Anonymous Coward · · Score: 0

    I'll stick with FairUse4WM, since the file that goes in is the same as what comes out, minus DRM. I'd rather not mess with unnecessary conversions, if possible.

    1. Re:thanks, but no thanks by kfg · · Score: 1

      Just show it where you store your DRM music. . .

      On a DRMed music free disc you can search forever.

      KFG

  6. Analog? by qbwiz · · Score: 1

    If it just uses the Windows mixer and the sound never actually leaves the soundcard, I suspect that it just stays digital the entire time, and is never actually converted to analog. I'm not sure how the Windows soundcard interface works, so I might be wrong. In any case, if you're using this program to play WMA files, you're still degrading their quality by transcoding them to MP3. That probably won't matter if you're just going to play them on your iPod though.

    --
    Ewige Blumenkraft.
    1. Re:Analog? by Anonymous Coward · · Score: 1, Interesting

      The wma is decoded to produce the sound signal that is played through your speakers. This program just recaptures the signal and encodes it as an MP3. It is no different than reencoding a file from one format to another, but you cannot reencode easily on DRM'd files.

    2. Re:Analog? by Joebert · · Score: 2, Insightful
      If it just uses the Windows mixer and the sound never actually leaves the soundcard, I suspect that it just stays digital the entire time, and is never actually converted to analog.

      I hope you're right, I get the feeling heads would roll if the general public found out the digital music stuff they sold a kidney for was just converting it back to what they already had before they actually hear it.
      --
      Wanna fight ? Bend over, stick your head up your ass, and fight for air.
    3. Re:Analog? by tolan-b · · Score: 1

      There is more degradation this way, because instead of having:

      decode->encode

      you have

      decode->adc->dac->encode

    4. Re:Analog? by qbwiz · · Score: 2, Insightful

      Right. So unless it's going through a DAC, it's the digital hole. Anyway, I thought everyone knew not to transcode files.

      --
      Ewige Blumenkraft.
    5. Re:Analog? by FLEB · · Score: 1

      Assuming you're not patching from the Line Out to the Line In with an analog cable (if you're just using the soundcard's "record wave" mode), would it be running it through the DAC or ADC?

      --
      Information wants to be free.
      Entertainment wants to be paid.
      You just want to be cheap.
    6. Re:Analog? by hankwang · · Score: 1
      the sound never actually leaves the soundcard, I suspect that it just stays digital the entire time,

      Correct. Unfortunately, most consumer-grade soundcards resample all channels to 48 kHz, which means that the 44.1 kHz data stream will be resampled two times: once from 44.1 to 48, and then from 48 back to 44.1. Although it is in theory possible to do that without change of the data (48 k should contain redundant data), in practice the re-sampling will introduce artifacts. Resampling well is especially computationally intensive if the difference in sample rates is so small.

      Anyway, these artifacts will probably be neglegible compared to the compression artifacts caused by the encoder. Especially mp3 (with LAME) does a very bad job dealing with compression artifacts from the previous encoding, even if the original encoding was a high bitrate. See also the hydrogenaudio.org wiki

    7. Re:Analog? by tolan-b · · Score: 2, Informative

      Well seeing as we've been told it's using the 'analog hole' I think it's a fair assumption that that's how it works.

      Seems the analogue in can capture the analogue out before it leaves the card, presumably bypassing whatever DRM enforcement happens in the lower level Windows Media layers:

      "Windows Media Player does the tough job of converting the 1's and 0's particular to that codec the music was stored as into an analog output that is played through the sound card. While the song is playing, AnalogWhole re-routes this analog signal back into the recording input of the sound card. "

    8. Re:Analog? by Wovel · · Score: 1

      Umm no one ever hears digital audio. At some point, no matter how much money you spend, it has to be converted to an analog signal so your ears can hear it.

  7. Allright but.. by The+Creator · · Score: 1

    Will it run on Vista?

    --

    FRA: STFU GTFO
  8. No WMA files here... by Anonymous Coward · · Score: 1, Insightful

    I don't have any WMA files or any other DRM crippled music on my computer, do you?

    1. Re:No WMA files here... by Potor · · Score: 1

      me neither - i don't imagine many /.rs do ...

  9. An alternative by moggie_xev · · Score: 2, Informative

    Look at http://www.highcriteria.com/ Total recorder when I was more windows centric I used it and I was happy.

  10. And this is new how? by msimm · · Score: 1

    Tunebite has been around for a while now (probably only one among many, but the only one I've actually used). It provides its own driver allowing accelerated encoding of both Window Media and iTMS files (video too, which is what got me interested, but doesn't seem to work as well, at least not with my temperament).

    --
    Quack, quack.
  11. PatchGuard by Constantine+XVI · · Score: 2, Insightful

    If it just pipes sound output from the mixer to MP3, what are the chances that Vista could block off access to mixer output except for low-level (driver) access, which is then blocked by PatchGuard?

    --
    "I think an etch-a-sketch with an ethernet port would beat IE7 in web standards compliance."
    1. Re:PatchGuard by Anonymous Coward · · Score: 0
  12. Spend 3 minutes naming it next time, not 2. by BeeBeard · · Score: 2, Funny

    I dare you to find anything at all funny about the word "AnalogWhole".

    1. Re:Spend 3 minutes naming it next time, not 2. by gameforge · · Score: 1

      Anna's Log Hole?

      You know what the ants standing on the turd in the toilet were singing? "When the log rolls over we're all gonna die..."

      I know. == !(that funny).

    2. Re:Spend 3 minutes naming it next time, not 2. by Joebert · · Score: 0, Redundant
      Other than me seeing "AnalogWhore" at first glance of the title, I would imagine years from now when virtual reality is common this word could be funny.

      Great, great, resurected Grandfather to great, great, test tube Grandson while gangbanging virtual hookers at digiHooker.cum : "Boy, when I was your age, we didn't have all this fancy shmancy DigitalHole stuff, why, I had to whip out my piss stick & put it in an AnalogWhole !"
      --
      Wanna fight ? Bend over, stick your head up your ass, and fight for air.
  13. well by Nasarius · · Score: 1

    They don't mention re-compression. If they're using the Apple lossless format, quality loss should be negligible unless you have a really awful soundcard.

    --
    LOAD "SIG",8,1
    1. Re:well by omeomi · · Score: 1

      They don't mention re-compression. If they're using the Apple lossless format, quality loss should be negligible unless you have a really awful soundcard.

      The Apple iTunes AAC format is not lossless. At the bitrate that they use for most of their stuff, it's not even close. Whenever you're going from one lossy compressed format (in this case AAC) to another lossy compressed format (MP3), there will be recompression. There's no other way around it.

    2. Re:well by Anonymous Coward · · Score: 0

      go back and read it again. "Apple lossless format" is not AAC. They are two different things.

      Don't take my word for it, go read about it at wikipedia.org/wiki/Apple_Lossless.

    3. Re:well by Anonymous Coward · · Score: 0

      Nobody said AAC is lossless. The grandparent was talking about the Apple Lossless Format.

    4. Re:well by Anonymous Coward · · Score: 0, Informative
    5. Re:well by ocelotbob · · Score: 2, Insightful

      Apple has a lossless codec in addition to AAC. It's playable in itunes and the ipod.

      --

      Marxism is the opiate of dumbasses

    6. Re:well by omeomi · · Score: 1

      Apple has a lossless codec in addition to AAC. It's playable in itunes and the ipod.

      I wasn't aware of that. However, it's somewhat irrelevant as the topic of the conversation related to transferring files from WMP playable formats into MP3. Unless Apple has been more open with their lossless codec than they have been with their version of AAC, it's doubtful that WMP is able to play the files.

    7. Re:well by Hal_Porter · · Score: 1

      The lossless codec has been reverse engineered. WMP uses ActiveX components to decode audio, so it's not that hard to add support for any audio format if you know how to write .ax dll. Other media players have their own codec formats, and in fact FFMPEG has support for ALAC, so you don't need to.

      --
      echo -e 'global _start\n _start:\n mov eax, 2\n int 80h\n jmp _start' > a.asm; nasm a.asm -f elf; ld a.o -o a;
  14. oops by Nasarius · · Score: 0

    Clearly, I'm an idiot and/or I can't read. Re-encoding as MP3 is a terrible idea.

    --
    LOAD "SIG",8,1
    1. Re:oops by Firehed · · Score: 2, Insightful

      Not as terrible as buying low-bitrate music with DRM was in the first place.

      --
      How are sites slashdotted when nobody reads TFAs?
    2. Re:oops by krell · · Score: 1

      Why, because MP3's don't play on commodore 64?

      --
      Where were you when the voynix came?
  15. Hey BeeBeard by Anonymous Coward · · Score: 0

    Have you taken your head out of your analogwhole yet?

  16. Rudy Van Gelder by Einstein_101 · · Score: 1

    With all this focus on the amount of quality lost in the reconversion, people are overlooking the most important issue:
     
      DRM'd music and .wma have mediocre sound quality to begin with.
     
    Being considered good quality on computer speakers or iPod ear buds is one thing; sounding good on $150 audiophile earphones or a dolby digital surround system is another thing entirely. I see this all the time with old Jazz records. You can re-encode with the best software modern science can provide, but it doesn't mean a damn thing if the initial conversion was a crappy one.

    1. Re:Rudy Van Gelder by chmod+a+x+mojo · · Score: 2, Interesting

      DRM'd music and .wma have mediocre sound quality to begin with


      Can you please justify this? I have a Klipshe ProMedia 5.1 surrounds system with an SB Audigy Gamer Edition (yes ancient sound card but it sounds beautiful to me) and I can not tell the difference between a high bit rate .WMA file (or even one of the "lossless" compression ones.) and the same song playing from CD. The Klipshe setup i have was also one of the "Kick Ass" rated setups from Maximum PC before they changed the styling and unfortunatly lost some sound quality in ~2002.
      Now don't get me wrong, maybe you DO hear a difference, but I don't and I have been a "audiophile" for many years, i can't listem to music unless it is on a HI-FI stereo in my car, same for my home theater system.
      I even remember a Maximum PC article ( lete 2005-ish) where they took a bunch of people and played music THEY brought in and had them try to tellthe difference between .wav 128K 296K and the 392K compression schemes (well it was on MP3 format but still the same goes) and most of the people got them wrong.


      I apologize if i rambled a bit i am posting from a hospital. They have free WIFI....woot! Now if only the painkillers were free :-(

      --
      To err is human; effective mayhem requires the root password!
    2. Re:Rudy Van Gelder by RonnyJ · · Score: 1
      DRM'd music and .wma have mediocre sound quality to begin with.

      The presence of DRM has absolutely no effect on the audio quality whatsoever - once it's decoded, the audio data is exactly the same as an equivalent file without DRM.

      You could say that the most common DRM music suppliers encode in a mediocre format (although many would challenge that), but the fact that they use DRM is irrelevant to the audio quality.

    3. Re:Rudy Van Gelder by PenGun · · Score: 1

      Klipsch is the word you are looking for. They have never made an accurate speaker. Even the mighty Klipsch horn based corner filling beasts from the 50s were anything but accurate, impressive is what they do..

        What you have is a poorly damped overly sensitive mush maker. It's not surprising you can't hear differences between compressed and merely digitaly simulated files, or between WMAs and WAVs if you like that better.

        I guess I have to pull out my c**k here to prevent bandwith abuse.

        M-Audio 24/96 Audiophile (the weakest link, the record player just destroys any digital effort BTW)
        Sonic Frontiers SFL1 Signature with gold pin Mullard (factory modded)
        Kimber Cable braided interconnects (cheap but very good)
        Sonic Frontiers SFM75 Monoblocks with Svetlyana 6550Bs
        Tara Time and Space Speaker Cables (oldies but goodies)
        B&W Matrix 1 Speakers bi-wired with crossovers rebuilt with Multicaps (best bang for $ I ever experienced in audio)

        The difference is obvious if you have the tools.

          PenGun
        Do What Now ??? ... Standards and Practices !

    4. Re:Rudy Van Gelder by radish · · Score: 1

      I can not tell the difference between a high bit rate .WMA file (or even one of the "lossless" compression ones.) and the same song playing from CD
      Well you're not supposed to be able to tell the difference with the lossless one, because it's, well, lossless. It is the exact same audio data as from the CD. The regular WMA one on the other hand is easily discernible with the right system and ears. If you can't tell the difference, great, that makes your life a lot easier :)

      --

      ---- Den ene knappen er powerknapp, den andre er Bender voice knapp "Bite My Shiny Metal Ass"

    5. Re:Rudy Van Gelder by slaida1 · · Score: 1
      Tara Time and Space Speaker Cables (oldies but goodies)

      Ha ha, cables! May I suggest you also try wooden knobs? ;)

      --
      Preserve old classics: copy your collection onto all hard drives.
    6. Re:Rudy Van Gelder by PenGun · · Score: 1

      Nah ... I kind of tread a middle path. I try to get the best I can for a reasonable $ and just normal physics is good enough for me ;). The cables are actually pretty nice with spaced wires for dielectric coherence. I dunno, I got em' pretty cheap and they do sound good.

        My favorite thing, to open a few people's eyes anyway, is to simply replace the interconnects from their CD player to their Preamp or Amp inputs with braided Kimber Cable, an amazingly cheap interconnect, and ask em' if it makes a difference. It almost always does noticably improve the sound.

          PenGun

  17. Secure Audio Path is in Windows ME, XP, and Vista by tepples · · Score: 5, Informative
    what are the chances that Vista could block off access to mixer output except for low-level (driver) access

    Very high. Windows Millennium Edition and Windows XP operating systems already support the Secure Audio Path, which places the (WHQL logo approved) decrypter, (WHQL logo approved) decoder, and (WHQL logo approved) audio output driver in kernel space. Part of the WHQL logo requirement is that no driver may mix Secure Audio Path audio into any cleartext digital output, and no driver without a logo is a valid Secure Audio Path playback device. However, few if any WMA files that require the Secure Audio Path are in the wild yet. However, record labels will begin to change their requirements as WMA stores' customers replace their computers that came with Windows 98 or Windows 2000 with newer computers that come with Windows Vista.

    For WMA files that use Secure Audio Path, you'll need a $5 audio cable and Audacity.

  18. Freeware programs by WhatDoIKnow · · Score: 1

    that capture anything going to the speakers from the sound card and save it as mp3 have been available for years. Maybe they don't get the metadata, but that wouldn't be too tough to fix by other means.

    :wq

  19. Would be more impressed with a digitalwhole by sokoban · · Score: 1

    DACs and ADCs and output stages on most soundcards are pretty awful. I would think that using a loopback of a digital audio out would be much better.

    --
    09 F9 11 02 9D 74 E3 5B D8 41 56 C5 63 56 88 C0 is the magic number.
    1. Re:Would be more impressed with a digitalwhole by SillyNickName4me · · Score: 1

      And on top of that, many a sound chip nowadays only does 48khz samplerate, which means you get some crappy resampling at least once, and probably twice.

    2. Re:Would be more impressed with a digitalwhole by tepples · · Score: 1
      DACs and ADCs and output stages on most soundcards are pretty awful.

      Then buy a USB or FireWire sound card. They generally have higher quality DACs and ADCs that sit outside the electrically noisy PC case.

  20. People are missing the point by Anonymous Coward · · Score: 0

    I suspect the most important point of this project is not to make MP3 from WMA, but to show the utter futility of DRM mechanisms.

  21. An MP3? by Fear+the+Clam · · Score: 2, Funny

    Shouldn't be a problem. Heck, you could even say that it plays for sure.

  22. The Anal Ogg Hole by Anonymous Coward · · Score: 2, Funny

    Good name for a pr0n flick about open source audio codecs, yes?

    Yes, I know what you're saying.. there aren't any porn flicks about open source software.

    I aim to change that.

    As soon as I get a video camera and work up the nerve to leave mom's basement. *peeks out window*

  23. DEAR SIR by Anonymous Coward · · Score: 0

    Your ideas intrigue me and I wish to subscribe to your newsletter.

  24. Quit Your Sniping and See the Benefits by Nom+du+Keyboard · · Score: 1
    Get off your high horses about already having this facility in some other, already existing, manner and see the benefits. This is another arrow in the quiver of those fighting DRM and the right to use your music as you wish. So what if there are other methods available. Some day those may be closed off, while this still works. There are a lot of people out there being paid to work full-time on shutting down every method of unlocking DRM for fair use.

    Anything that shows the futility of the whole idea of DRM is a good thing!

    Anything that may still work the day everything else stops working is a good thing!

    Anything that makes their job harder by forcing them to divert their efforts to yet another hole in the dike to plug is a good thing!

    So quick being fsking pseudo-geek snobs and rejoice that yet another method has been found.

    --
    "It's the height of ridiculousness to say for those 9 lines you get hundreds of millions."
    1. Re:Quit Your Sniping and See the Benefits by Stormwatch · · Score: 2, Interesting

      Yes, but if you have those DRM'ed files, it means you have bought them. Your dollars told the record company that you accept DRM, even if you find a workaround later. Of course, it is a good thing that this workaround exists; but, as a principle, one should not have bought that junk in the first place!

    2. Re:Quit Your Sniping and See the Benefits by linefeed0 · · Score: 1

      I'm not so sure that this is a good thing at this stage. It's less of "another hole in the dike"; if people use this and spend their time on it, that is less time spent on cracking the DRM where it really hurts. It almost seems like a flag of surrender on the DRM issue, and it would be better to create tons of uncertainty and doubt that DRM works at all (in the digital, compressed original) by repeatedly cracking it wide open than to make media companies think that we're resorting to this because we can't manage or are too lazy to crack the original. This could cause them to redouble their efforts to plug the analog hole or make using it circuitous and obnoxious.

    3. Re:Quit Your Sniping and See the Benefits by MooUK · · Score: 1

      The issue is more that this particular method is nothing new either. It's NOT another new way of doing things; it's an old and common method in new clothing.

    4. Re:Quit Your Sniping and See the Benefits by Lewrker · · Score: 1
      You are mistaken. It is still the very same method.

      Anything that shows the futility of the whole idea of DRM is a good thing!

      It doesn't show the futility of anything. The music you get in result is reencoded, that means really bad quality. So you have to break your music in order to exercise your rights as a customer.

      Anything that may still work the day everything else stops working is a good thing!

      What do you mean ? In the new uber extra high-definition quality anything standards everything gets encoded even on the way from one device to another. And we still haven't got a method to crack DRM in any other way.

      Anything that makes their job harder by forcing them to divert their efforts to yet another hole in the dike to plug is a good thing!

      Divert their methods ? I believe they have been using the same methods for some several years now. Your comment is totally irrelevant. Nothing really get's cracked or bypassed here. It's just the magic of analog music. It's about the same as learning the song by heart and then rewriting it and performing it. Might work. But it's not the real thing. And you still didn't crack anything.
  25. Sigh. by daeg · · Score: 2, Insightful

    What can be seen or heard can be copied, no matter how difficult you make it.

    1. Re:Sigh. by fizzup · · Score: 1

      When I first heard of the zero knowledge proof of knowledge, it really bent my mind. But DRM authors are trying something truly astounding: the zero knowledge transfer of knowledge. I wish them good luck.

  26. Analog Loopback by nurb432 · · Score: 1

    Isnt that soon to be disabled/removed due to DRM/attorneys ?

    --
    ---- Booth was a patriot ----
  27. Good method for most. by Programit · · Score: 1

    Contrary to the knockers and music perfectionists, this method works very well. I've been using a similar method for years and to be honest, the quality loss is that small that only the perfectionist in ideal conditions will pick it up. 99% of the people couldn't tell the difference in normal surrounds! Well Done!

  28. If you don't want to lose quality... by gregorio · · Score: 4, Interesting

    ...build your own USB "converter". Companies like Texas Instruments have lots of devices like PCM2704, that allow access to an unprotected sound bitstream. It's pretty simple to build a fake digital speaker that just redirects the data to a fake digital line in. Some microcontrolled usb sound devices contain both input and output devices on the same IC, so you can software redirect the output (coming from the computer) to the input (going back to it).

    So you don't even need an "Analog hole". You can use a digital hole and don't lose any quality at all. And this kind of device is perfectly accepted by any "content protection" driver schemes.

    It's impossible to protect sound files.

    1. Re:If you don't want to lose quality... by Anonymous Coward · · Score: 1, Informative

      There is also a purely software-based solution that doesn't lose quality: QEMU. Install this emulator, instal Windows inside there, install drivers for the emulated SB PCI sound card (they already have the needed signature), and redirect the emulated sound output to a wav file. You'll get a bit-precise copy of the sound.

    2. Re:If you don't want to lose quality... by evilviper · · Score: 1
      It's pretty simple to build a fake digital speaker that just redirects the data to a fake digital line in.

      Umm... it's even simpler to connect the digital out to the digital-in on my current soundcard.

      You can use a digital hole and don't lose any quality at all.

      This is just WRONG. You will still very likely lose some quality due to sampling rate conversion your soundcard automatically does.

      And besides that, we're talking about re-encoding to MP3 afterwards, so the D/A and A/D conversion with a decent soundcard will be absolutely insignificant, next to the MP3 re-encoding. Only with lossless audio codecs would this matter at all.
      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    3. Re:If you don't want to lose quality... by Anonymous Coward · · Score: 0
      This is just WRONG. You will still very likely lose some quality due to sampling rate conversion your soundcard automatically does.
      You appear to mistake half a decade old soundblaster live cards with real soundcards with digital in and out. The latter can be distinguished from the former by the complete absence of a nonsenical hardware mixer that forces its sampling rate conversion on everything. It may be news to you, but there are even US$30 USB sound"cards" with digital in and out that don't modify the bitstream. Mine can flawlessly play and record DTS and AC3 audio at 44kHz and 48kHz, which is a pretty clear indicator that nothing gets resampled.
    4. Re:If you don't want to lose quality... by gregorio · · Score: 1
      There is also a purely software-based solution that doesn't lose quality: QEMU. Install this emulator, instal Windows inside there, install drivers for the emulated SB PCI sound card (they already have the needed signature), and redirect the emulated sound output to a wav file. You'll get a bit-precise copy of the sound.
      Yes, but that won't cover sounds encoded with the latest DRMed files that requires only "Trusted Computing" hardware and drivers. These kind of protections are embedded inside the O.S. and will not allow a "unsafe" sound path.
    5. Re:If you don't want to lose quality... by Bazzargh · · Score: 1

      There is also a purely software-based solution that doesn't lose quality: QEMU. Install this emulator, instal Windows inside there...

      Vista, doesn't permit access to DRM'd content if you run it virtualized. At least, thats the license requirement, and there is a a suggestion that they use red pill techniques to detect virtualization for runtime checks. (I don't use vista, and I don't own any DRM'd content, so I'm not commenting from personal experience)

    6. Re:If you don't want to lose quality... by gregorio · · Score: 1
      Umm... it's even simpler to connect the digital out to the digital-in on my current soundcard.
      No, it's not, as the new generation of Trusted Computing DRM will force the creation of a "Secure Audio Path". So your current soundcard will not be able to play files with the latest DRM and trusted cards will obviously include some kind of protection on the digital out bitstream.

      So if you simply connect out-in on a Trusted Computing Hardware, you'll not be able to record the file.

      This is just WRONG. You will still very likely lose some quality due to sampling rate conversion your soundcard automatically does.
      No, it's not wrong. The mentioned USB IC will not perform any kind of conversion. That's very computationally expensive for a US$ 3,00 integrated circuit. As another poster said, this kind of resampling strategy is used only by nonsense projects like some SB Live cards. DAC chips doesn't care about the conversion clock and well made digital sound processors can be programmed to accept generic sampling rates, and that's what most cards do these days, instead of wasting processing power with sample interpolation.

      And besides that, we're talking about re-encoding to MP3 afterwards, so the D/A and A/D conversion with a decent soundcard will be absolutely insignificant, next to the MP3 re-encoding. Only with lossless audio codecs would this matter at all.
      Insignificant or not, it's still worse than a pure-digital electronic solution.
    7. Re:If you don't want to lose quality... by caseih · · Score: 1

      Still loses quality, though. The significant quality loss is in the codec mainly, not necessarily signal losses, although this digital method would be of a somewhat higher quality than analog.

    8. Re:If you don't want to lose quality... by evilviper · · Score: 1
      No, it's not, as the new generation of Trusted Computing DRM will force the creation of a "Secure Audio Path". So your current soundcard will not be able to play files with the latest DRM and trusted cards will obviously include some kind of protection on the digital out bitstream.

      Your argument eats itself...

      When the switch to "Trusted" computing happens, you aren't going to be able to find any signed drivers for your USB device anyhow, so no output for you.
      --
      Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
    9. Re:If you don't want to lose quality... by gregorio · · Score: 1
      Your argument eats itself...

      When the switch to "Trusted" computing happens, you aren't going to be able to find any signed drivers for your USB device anyhow, so no output for you.
      It already happened (Secure Audio Path exists since Windows Me) and yes, simple USB speakers are supported by it. This kind of DRM scheme is not made to stop the inevitable, but to eliminate the trivial aspect of music sharing.

      No DRM scheme in the planet is going to ignore digital speakers, as most loudspeakers in the future will have a 100% be digital interface. But you'll need special hardware to get the precious bits out of these interfaces. Maybe not the most expensive hardware in the planet ($10 will always do it), but you'll have to build it yourself, as "music copying USB dongles" will experience the same litigation faith as videogame modchips.

      In the end, DRM is about making it difficult, not making it impossible.
  29. Time loss, not quality loss is the problem by metaphorever · · Score: 1

    No, the real issue that everyone is ignoring has nothing to do with audio quality or loss of it. The problem with these "analog hole" solutions is TIME loss. Seriously, if it's reading from the sound card it has to play each and every file, and for any reasonably sized music collection that will take a lot of time. Even with the added step of going from unDRMed WMAs to MP3s, FairUse4WM is infinitely faster because you can do everything in batches. This is a nice proof of concept, but if you really have a WMA music collection FairUse4WM is the only practical solution.

    --
    If people continue to abuse this feature, I will have to remove it. - Slashdot Comment Box, 1998
  30. OR.... by Anonymous Coward · · Score: 0

    You can buy a CD and never have to deal with DRM or sound quality loss.

  31. Parent is on the money by PopeRatzo · · Score: 1

    I was just thinking that I've been doing something very similar with an old PIII with an SBLive card. My daughter brings all these files that have been ripped from who-knows-where and wants them on her player. It works just fine and the sound is great, especially when we're talking about the music she listens to. It would take a lot of digital crud to make this stuff any less listenable.

    Seriously, I've seen old SBLive's with digital outs at the neighborhood used computer-gear store for 10 bucks. Good, stable drivers are readily available and some of them even have these nifty breakout boxes with optical in-outs.

    Now, if I could just find a ready supply of PCI sound cards with old-fashioned game ports I can use for midi I/O. I've got a crraaazy project in mind, but it seems that all the new two-buck sound cards on the market don't do the game-port/midi thing any more. Just audio ins and outs. For what I want, a USB midi adapter won't do, I need the old fashioned sound card with game port, and about 6 of those in PCI.

    --
    You are welcome on my lawn.
    1. Re:Parent is on the money by Anonymous Coward · · Score: 0

      First of all, I am TERRIBLY SORRY for posting wildly off-topic. Unfortunately, you closed replies to your journal, and this is the only way I can tell you this.

      Your journal entry about people smoking - first of all, I smoke (a LOT) and really, really wish it wasn't harmful. But, as my mom is dying of C.O.P.D. and Emphysima (Goddam dude, I've cried hard over this) I really wish you'd reconsider telling people that there's nothing wrong with it.

      I know the whole correlation/causation thing. I can't actually prove that smoking caused her condition. However, she does go to the best respiratory/pulmonary hospital in the country (National Jewish in Denver, CO), if not the world, and they seem to think it's exactly what caused it. I'll take their science over the EPA's any day, that's for certain.

      I almost lit up when I read your journal, like "Hey, what's this? Someone found out that I'm not killing myself after all?" But as soon as I remember my mom, I mean Jesus. I can't believe that.

      Again, very sorry for posting so off-topic.

    2. Re:Parent is on the money by Anonymous Coward · · Score: 0

      Sound Blaster Live!, dude- cheap, PCI, and readily available via E-bay.

    3. Re:Parent is on the money by PopeRatzo · · Score: 1

      Oh man, I didn't mean for anybody to read that. I thought journals were just places to put private stuff, sort of alike a personal clippings box.

      The reason I copied those links about smoking into my journal is so I could remember to check the accuracy of that stuff. I sure don't believe any of the claims that smoking isn't deadly.

      Those links were posted by someone else after I'd made a comment that slammed the tobacco industry. They weren't mine, and I didn't mean to spread that disinformation to others.

      I'll take down that journal immediately, and I'll take the time to find out what Journals are really for.

      And I'm really sorry about your Mom. I detest the fact that there are industries in the world that will poison their customers just to increase profits.

      --
      You are welcome on my lawn.
    4. Re:Parent is on the money by gameforge · · Score: 1

      Hey,

      I wrote that anonymously because the story was still kind of active and I didn't want people to have to read a bunch of off-topic stuff.

      That makes perfect sense to me; I (truly) appreciate your condolences for my mom. Wish me luck! I run out of cigarettes next week, and hope to stop cold turkey, though I have nicotine gum & some old patches just in case I need help...

      ~ Your friend gameforge

  32. Not with good files. by lindseyp · · Score: 3, Informative

    Most people who say this are used to mp3s being low quality. I too can easily tell the difference when quality is that low.

    But for "LAME --preset insane" quality files, which tend to be about 2x the filesize, I've done my own blind tests on high end equipment: i.e.:

    Winamp

    ->Audiophile24/96 sound card

    -> Benchmark DAC1

    -> Decware Zen Triode Integrated Amplifier

    -> Gallo Nucleus Reference II speakers

    Or replace the DAC and amp with a Denon AVC-A1SE amplifier (that's a ref. quality $5000 a/v amp)

    I've also listened with Sony MDR D77 headphones, and Shure E3 studio monitor earphones with both of these amps.

    In my own conclusion I couldn't tell the difference.

    I coded the files back to WAV, a mix of high quality recordings of classical, rock, techno and Clapton, and invited a self-professed bunch of audiophiles to volunteer their opinion on which were the true WAVs and which had gone through the mp3 coding process. Nobody volunteered an opinion.

    Since then I always code my music to mp3 using that setting. I've DJd using that quality of file with Virtual DJ with no pitch correction (important, this affects quality a lot) and had other DJs tell me they couldn't believe I was not using Vinyl.

    I wish I still had the files I prepared, I would post them here for your enjoyment, but I don't doubt some slashdot genius would come back with the correct answers by examining the files digitally.

    --
    j'ai découvert une démonstration vraiment admirable (de ce théorème général) que cette si
  33. Re:biZnAtch by CleverBoy · · Score: 1

    Yep. Every since I read the title of this article, this is image has been wallpapered to the back of my mind. Sigh.

  34. Dont know much about law by cky625 · · Score: 1

    Dont know much about copyright law, if i'm wrong please correct me. Here is a some action, could be music/movie/whatever. There whole action is being captured, processed and finally into a digital media. The media, which is different from the action, could be reproduced identically in digital form, meaning that it is unauthorizely manufactured. Same could apply onto books distrubuted in plain letter size paper form being photocopied, in college it's serious. What is permitted in school is that you learn from that book and write your own version, plagiarism lies beeneth whatever you are just simply memorizing it and then rewrite in your paper or you abosrb the knowledge and have altered (e.g. add/subtract) the main concept and wirte somthing different(proof is alter also since its more proofable). When paper has become punch card and to hard disc now, brain proccess become recode, then loop back and record sounds way dark then in the grey area.

  35. Now the DMCA has a new enemy by 1310nm · · Score: 1

    The too-lazy-to-put-a-loopback-cable-on-their-soundcar d teenagers with too much time on their hands.

    1. Re:Now the DMCA has a new enemy by MooUK · · Score: 1

      As I understand it, this particular method doesn't get as far as being converted to analog sound; it's more just redirecting the audio stream to a file. Which isn't anything new/

  36. DAC/ADC is not the loss source -- transcoding is. by Dr.+Zowie · · Score: 1

    DAC and ADC circuits are really good these days. By really good I mean that a $100 sound card is better than a high-end tape deck from the 1980s, or even than most audiophile turntables playing brand-new vinyl. The built-in soundcard on your motherboard probably "sucks", which means it's only as good as that really nice component tape deck your older brother bought in the 1990s and you drooled over until you discovered mp3s. The suckiness is probably digital noise from the motherboard, leaking it at the -50 or -60 db level (about the same as the noise floor for a cassette tape w/o Dolby or DBX). Harmonic distortion is probably buried in the digital leakage, even on cost-engineered, sucky on-board sound.

    A few years ago I did audio comparisons between a cheap-ass I-Opener computer playing mp3s ripped from a record and a midrange Technics tape deck playing the same tracks recorded from the same record, and the I-Opener did better.

    So if you've paid any attention at all to your sound card, you probably won't hear any distortion from passing the sound through it. You're much more likely to notice the fuzz and tinkle-bells from the initial low-quality Rhapsody encoding.

  37. Re:biZnAtch by Foobar+of+Borg · · Score: 1
    Yep. Every since I read the title of this article, this is image has been wallpapered to the back of my mind. Sigh.


    And to think all this madness started with the ancient Chinese Taoist Goa Tse.

    And hopefully the humorless bastards at Wikipedia haven't deleted this yet.

  38. Virtual Audio Cable - Old hat by flyingfsck · · Score: 1

    Old hat:
    http://software.muzychenko.net/eng/vac.html

    of course, on Linux there has always been sox, which is even older hat:
    http://sox.sourceforge.net/

    --
    Excuse me, but please get off my Pennisetum Clandestinum, eh!
    1. Re:Virtual Audio Cable - Old hat by sillybilly · · Score: 1

      There is gonna be more legislation forbidding workarounds to DRM, including using a microphone or camera and unknowiningly recording copyrighted material will carry full punishment. Safest thing to do is to never click copy and never record anything, because after copyright is fully instituted, da man will take over all recording rights too, unless you pay a fee, to get a recording license just like a drivers license, and pass a written exam. After all recording stuff is just as dangerous as driving, if not more!

  39. Shouldn't this be called aWhole for short? by Anonymous Coward · · Score: 0

    Shouldn't this be called aWhole for short?

  40. done this since 1991 by emptybody · · Score: 1

    with the internal audio header cable from the CDROM drive to the motherboard.
    play music record to datafile from audio in.
    no microphone involved.

    --
    comment directly in my journal
  41. DRM - Services vs. Purchases by insaneshow · · Score: 1

    I tend to get in a lot of arguments over DRM. Simply because I am against DRM, yet I subscribe to Urge. However, my concept is pretty simple; If I buy music, I do not want to be restricted in how I can use it. If I subscribe to a service, I expect that there will be restrictions in use. In growing up I have come to see that downloading music is indeed stealing, regardless of how corrupt the music industry is. By subscribing to a service like Urge, I have been able to weed out more crap music and more effectively spend my music dollars. My whole argument boils down to purchase vs. service. All that being said, I first got really excited when I read this article. I thought of listening to all of those new albums that I've been listening to on my Creative player on my much preferred iPod. I even went as far as downloading the program. But after it sat with me for a few moments, I thought that this was no better downloading tracks off of a torrent. Simply because I did not pay for the tracks, but for the service. Had i purchased those tracks, I would have full control of them. I will only purchase music in a physical form. Atleast those are my thoughts on the subject. ~InsaneShow

    1. Re:DRM - Services vs. Purchases by Anonymous Coward · · Score: 0

      Die in an inferno.

  42. Could be done years ago by Anonymous Coward · · Score: 0

    Open the song in a wav/mp3 tracker, export or play and capture the input from the line out. Easy except now you have a nice raw file of a mp3, then recompress it to mp3 you'll get 128kbps or whatever you use of a already compressed file.

  43. virtualization/emulation by BerkeleyDude · · Score: 1

    Suppose you run Windows in VMWare or some other virtual machine or emulator. Use Linux as a host, and intercept the sound there. Signed device drivers, secure path, etc. don't matter any more, do they?

    If you can hear the sound, you can save it. Microsoft cannot do anything to stop you.

    (Of course, they can try to detect VMWare and refuse to play any sound. But: 1) that will break lots of legitimate uses of Windows, and 2) you can always make an emulator that looks just like real hardware.)

  44. MOD PARENT UP by Anonymous Coward · · Score: 0

    +5 - Insightful.
    No, seriously. Will we be able to do this kind of thing on the 'next genration/s of Windows (tm) Software'?
    If **AA get their way then, possibly.. no.

    Word of the day: Mileage. How appropriate. My current hardware will hopefully last me a while. Maybe.

  45. iTunes != DRM by daBass · · Score: 1
    Uninstall iTunes or whichever thing is giving you this unplayable worthless crap to begin with

    When did iTunes ever give you "unplayable worthless crap"? It's the iTunes Store that sells you that. iTunes the application merely provides a way to play it back. Nothing you rip yourself in iTunes has any DRM on it whatsoever.

    Just to set the record straight...

    1. Re:iTunes != DRM by gameforge · · Score: 1

      I don't use iTunes. I know there are gazillions of MP3 player software packages, and if DRM didn't exist and everyone sold plain unDRM'd MP3s, you wouldn't need the iTunes player, except maybe for getting your MP3s onto your iPod... I really don't know how that works, since I don't own an iAnything.

      Clearly, you don't HAVE to use iTunes to rip your music; I was referring to the AAC DRM files that require it to be played (if I understand it correctly).

      iTunes doesn't work with Linux because of its DRM, right? iTunes is garbage. Buy non-DRM MP3's from somewhere else, and use an MP3 player that doesn't give a flip, and another of the gazillion CD ripping programs (is roughly what I was trying to get across, if you have absolutely no way to play your DRM files the way you want).

      I blame Sony more than iTunes for DRM. But iTunes is where all these teenagers I hire get this DRM junk that doesn't play nice with Linux. So whether it has any legit uses besides interfacing with the iPod, it goes into the useless junk software category for me.

      Please note: I don't intend to be offensive to people that use DRM media, or who use iTunes... I'm always on your side, unless you actually agree with DRM and condone it.

    2. Re:iTunes != DRM by daBass · · Score: 1

      I still don't think you get it: music ripped in iTunes does not have DRM on it. I have an iPod and use iTunes all the time. But I have no DRMed music at all; everything is ripped off CDs I own. I now rip to ALC (Apple Lossless) but my wife uses 128K AAC because she only has a 4GB player. Again, these AAC files have NO DRM on them; only files *bought* from the iTunes Store are DRMed.

      iTunes doesn't work on Linux because Apple hasn't bothered porting it to it; the DRM has nothing to do with it. (if they did decide to port it, it obviously would have the DRM features for music bought from their store, but I guess selling DRMed music to your average linux user is like selling sunscreen to eskimos!)

      Until someone starts selling unprotected major-label music in a high bitrate, I'll stick with CDs. No DRM to deal with and if you know where to shop, they are even slightly cheaper than DRMed and terrible sounding 128K AAC files...

    3. Re:iTunes != DRM by gameforge · · Score: 1

      I'm sorry, then my comment doesn't apply to you.

      I get it.

      I'm talking about people who use iTunes (the STORE) to purchase their DRM music instead of another source, and who are forced to use iTunes (the PLAYER) to play it, and wish to convert it to a non-DRM format like the article & summary are about. I know people who use iTunes to buy music and don't even own an iPod.

      I understand iTunes can rip CDs to nonDRM MP3s and put them on your iPod for you. I knew this when I first posted. This is a legit excuse for this software to exist.

      Sorry if I confused anyone.

  46. Anal...log...hole by Anonymous Coward · · Score: 0

    You mean other than the fact that it contains the words "anal" "log" and "hole"? Nope, nothing at all questionable about it. [whistles]

  47. DMCA 1201 violation? by Aaarrrggghhh · · Score: 1

    The DMCA does have that nasty 1201 section which makes tools of circumvention illegal.

    No person shall manufacture, import, offer to the public, provide, or otherwise traffic in any technology, product, service, device, component, or part thereof, that - (A) is primarily designed or produced for the purpose of circumventing a technological measure that effectively controls access to a work protected under this title;

    But that's just for the US. Everyone else, enjoy.

    1. Re:DMCA 1201 violation? by dastardly_villain · · Score: 1

      So I send my hard drive to Estonia and have my friends there do it for me. Great problem solved. =)

    2. Re:DMCA 1201 violation? by pruss · · Score: 1

      IANAL. But the point of this seems to be that the access to the protected data is duly authorized, since an authorized media player is used. The authorized media player then decodes the data. It is the decoded data that is captured. The decoded data is not protected, and hence the DMCA may not apply to it (but don't trust me, ask a lawyer).

      Pocket DVD Studio apparently does this for DVDs.