AnalogWhole, an Alternative To FairUse4WM 168
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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well (Score:2)
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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.
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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.
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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.
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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>
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Exactly
Still loss of quality (Score:5, Informative)
Re:Still loss of quality (Score:4, Insightful)
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.
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Re:Still loss of quality (Score:5, Interesting)
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.
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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.
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Not with good files. (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
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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.
Parent is on the money (Score:2)
Seriously, I've seen old SBLive's with digital outs at the neighborhood used computer-gear store for 10 bucks. Good, s
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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.
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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
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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.
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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
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Okay 95% of everyone won't claim that their ga
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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.
It will TRADE that delay for OTHER artifacts. Typically, discarding MUCH more of the audio, and now having the artifacts of both audio codecs.
It isn't.
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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.
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A trivial and obvious typo. But troll away.
Your ears (or perhaps audio equipment) suck. Not everyone else's does.
And? I've been doing PROFESSIONAL studio work for a very long time now.
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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.
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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.
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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
iTunes != DRM (Score:2)
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...
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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?
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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
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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
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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.
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I'm not saying you are necessarily wrong, but according to the documentation for my lame (version 3.96.1):
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I always get the numbers backwards... it is -Q 0. You'd think the highest quality would be -Q 9, but no...
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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
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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 comp
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Yeah, it's Q=0.
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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
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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 ca
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From your source, regarding digital loopback @ 48KHz:
Re:Still loss of quality (Score:5, Funny)
mmm...but just listen to that lovely analog warmth! I'll take that over digital accuracy anyday...
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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.
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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
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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/ADC is not the loss source -- transcoding is. (Score:2)
And the point is?? (Score:2, Interesting)
So... (Score:2, Insightful)
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Analog? (Score:2)
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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.
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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
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decode->encode
you have
decode->adc->dac->encode
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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-
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Allright but.. (Score:1)
An alternative (Score:2, Informative)
And this is new how? (Score:2)
PatchGuard (Score:2, Insightful)
Secure Audio Path is in Windows ME, XP, and Vista (Score:5, Informative)
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)
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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).
Would be more impressed with a digitalwhole (Score:2)
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An MP3? (Score:3, Funny)
The Anal Ogg Hole (Score:2, Funny)
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*
Quit Your Sniping and See the Benefits (Score:2)
Anything that shows the futility of the whole idea o
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Sigh. (Score:3, Insightful)
Analog Loopback (Score:2)
If you don't want to lose quality... (Score:5, Interesting)
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.
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Umm... it's even simpler to connect the digital out to the digital-in on my current soundcard.
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
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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.
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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.
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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
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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
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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 [invisiblethings.org] 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)
done this since 1991 (Score:2)
play music record to datafile from audio in.
no microphone involved.
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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
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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
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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.
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And to think all this madness started with the ancient Chinese Taoist Goa Tse [uncyclopedia.org].
And hopefully the humorless bastards at Wikipedia haven't deleted this [wikipedia.org] yet.
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Pocket DVD Studio apparently does this for DVDs.