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AnalogWhole, an Alternative To FairUse4WM 168

Posted by kdawson from the digital-is-so-XX-century dept.
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."
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AnalogWhole, an Alternative To FairUse4WM

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    • Re: (Score:3, Funny)

      by kiwimade ( 891424 )
      Considering its targeting stuff like ipod playback, this shouldn't be a problem.
    • 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.

      • Re: (Score:2)

        by omeomi ( 675045 )
        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.

        • Re: (Score:2, Insightful)

          by ocelotbob ( 173602 )
          Apple has a lossless codec in addition to AAC. It's playable in itunes and the ipod.

          • Re: (Score:2)

            by omeomi ( 675045 )
            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.

    • Re: (Score:2)

      by geeber ( 520231 )
      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.

      • Re: (Score:3, Insightful)

        by Metteyya ( 790458 )
        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 autom
        • 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/productdet a ils.aspx?ObjectId=758&ParentId=114rel=url2html-241 56 [slashdot.org]http://www.rolandus.com/products/productdetails. aspx?ObjectId=758&ParentId=114>

          • 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
            • 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.
        • 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 cor

  • Still loss of quality (Score:5, Informative)

    by amplusquem ( 995096 ) on Saturday October 28, 2006 @05:44PM (#16625352)
    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 [hymn-project.org] which doesn't lose any sound quality.

    • Re:Still loss of quality (Score:4, Insightful)

      by omeomi ( 675045 ) on Saturday October 28, 2006 @06:04PM (#16625546) Homepage
      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.

    • Re: (Score:2)

      by westyvw ( 653833 )
      If your starting with crap like WMA then moving it to lesser but still crap MP3 why would you care?

      • Re:Still loss of quality (Score:5, Interesting)

        by gameforge ( 965493 ) on Saturday October 28, 2006 @07:24PM (#16626242) Journal
        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.
        • 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 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 t

          • Not with good files. (Score:3, Informative)

            by lindseyp ( 988332 )

            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

        • 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.

        • 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, s

        • Re: (Score:2)

          by tlhIngan ( 30335 )
          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

        • Re: (Score:2)

          by westyvw ( 653833 )
          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.

        • 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 SOUN

          • 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 ga

            • 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 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 b

                • 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

                  • 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 homin
        • 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...

          • 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?

            • Re: (Score:2)

              by daBass ( 56811 )
              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 de
              • 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 ex

        • Re: (Score:2)

          by smcdow ( 114828 )
          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.

        • Re: (Score:2)

          by shaka ( 13165 )

          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 <= qu

          • 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...

        • Re: (Score:2)

          by GWBasic ( 900357 )

          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 manipul

    • Re:Still loss of quality (Score:5, Funny)

      by Threni ( 635302 ) on Saturday October 28, 2006 @06:12PM (#16625614)
      > 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...
    • 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 y
      • 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.
    • 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.

      • 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

    • 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.

    • 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
  • 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.

  • So... (Score:2, Insightful)

    by Constantine XVI ( 880691 )
    So, it's just like using Audacity to record whatever goes through the sound card?
  • 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.

    • Re: (Score:2, Insightful)

      by Joebert ( 946227 )
      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.

    • 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 compu

  • Will it run on Vista?

  • An alternative (Score:2, Informative)

    by moggie_xev ( 695282 )
    Look at http://www.highcriteria.com/ [highcriteria.com] Total recorder when I was more windows centric I used it and I was happy.
  • 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).

  • PatchGuard (Score:2, Insightful)

    by Constantine XVI ( 880691 )
    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?
    • 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 [google.com], 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.

  • Spend 3 minutes naming it next time, not 2. (Score:3, Funny)

    by BeeBeard ( 999187 ) on Saturday October 28, 2006 @06:03PM (#16625528)
    I dare you to find anything at all funny about the word "AnalogWhole".
    • 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).
  • 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.
    • 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.

  • An MP3? (Score:3, Funny)

    by Fear the Clam ( 230933 ) on Saturday October 28, 2006 @06:27PM (#16625758)
    Shouldn't be a problem. Heck, you could even say that it plays for sure.

  • The Anal Ogg Hole (Score:2, Funny)

    by Anonymous Coward
    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*
  • 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 o

    • Re: (Score:3, Interesting)

      by Stormwatch ( 703920 )
      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!
    • 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 m

    • Re: (Score:2)

      by MooUK ( 905450 )
      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.

  • Sigh. (Score:3, Insightful)

    by daeg ( 828071 ) on Saturday October 28, 2006 @07:02PM (#16626064)
    What can be seen or heard can be copied, no matter how difficult you make it.
  • Isnt that soon to be disabled/removed due to DRM/attorneys ?

  • If you don't want to lose quality... (Score:5, Interesting)

    by gregorio ( 520049 ) on Saturday October 28, 2006 @07:51PM (#16626500)
    ...build your own USB "converter". Companies like Texas Instruments have lots of devices like PCM2704 [ti.com], 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.

    • 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 soundc

      • 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

        • 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.

          • 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 w

    • Re: (Score:2)

      by caseih ( 160668 )
      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.
  • with the internal audio header cable from the CDROM drive to the motherboard.
    play music record to datafile from audio in.
    no microphone involved.

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