Slashdot Mirror

It's true that a digital recording can never contain the amount of data in a vinyl groove, but who is saying that all the data in a vinyl groove is more of an accurate representation of all the data extant in the original sound wave than a digitally sampled recording?

The kicker for me showing a total lack of understanding of the technology is the popularity of USB turntables. They can't keep them in stock. Quick, someone show me any analog signal in a USB specification.. Analog is better.. Analog is king, Here use this USB turntable to enjoy your analog sound. What are they smoking? Nothing out the USB port of a turntable is analog in any shape or form. Who has a better low noise analog to digital converter, a consumer grade turntable or a CD mastering house?

Analog is king only because the mastering house slaughtered the conversion in the loudness war. If you check the links, the youtube link provides the best summary with an example of the problem which can be heard and seen.

http://createdigitalmusic.com/2007/05/16/loudness-war-music-over-compression-demonstrated-on-youtube/
http://my.opera.com/swerfot/blog/2007/08/26/loudness-war
http://en.wikipedia.org/wiki/Loudness_war
http://www.hydrogenaudio.org/forums/index.php?showtopic=55892

CDs are on the way out because the music on them is crap. Finding a decent recording in the pile of crap is why many simply avoid the contaminated format. USB turntables, even though you don't get analog, you also don't get the over compression, which is why the ability to play better source material is so popular. Analog has nothing to do with this argument. Destruction of the sound on compact discs in mastering is the problem.

None to my satisfaction, of course... but they do exist, some discussion in the hydrogen audio forums.

It's not "elitist fuckery", it's a combination of good equipment, an original source recording of high quality, and perhaps better-than-average hearing.

Every actual study I've seen shows that except on certain "hard" classes of recordings it's wishful thinking or other psychology--with 99% of normal music (be it well-recorded classical, a capella voice, rock, whatever) the 256kbps VBR LAME settings Amazon uses haven't been distinguishable to anyone in any scientific study I'm aware of.

I've had this argument many times, and there are some recordings that reveal obvious flaws in even 320 kbps CBR mp3 to my ears with my headphones and amplifier

If you can do it regularly, then you should pretty easily be able to make thousands of dollars for a few hour's work by claiming any of the numerous prizes offered for people who can succesfully distinguish 256kbps VBR MP3 of a wide array of popular music from uncompressed originals in a double-blind A/B test.

Now, if you're just talking a handful of special-case horrible-for-mp3 recordings (like, say, the well-known Eig "LAME killer" sample) then that's another story; some people can certainly pick that one out at 320 kbps VBR.

You can easily start off using a free ABX program like PC ABX (Windows) or LinABX, or a more expensive hardware solution. Just see if you can actually ABX them at home and if so, you should be good to go claim some cash.

It's pretty fun to see what you can actually distinguish, too. I have a nice setup with a good pair of Grados; out of my library of 4000+ songs there are maybe 3-4 I've found so far that I can pick out a 192kbps Ogg from FLAC. Before doing such testing, I was "sure" I could pick out the difference between the original and the 320kbps encodings I usually make. Most people who come over to my place can't distinguish 96 kbps from uncompressed except on a handful of nasty test samples, but if you actually learn what the common artifacts are it gets a bit easier for some people.

If I understood that correctly, eMusic is using LAME V0 VBR and Amazon uses LAME 256 kBit/s ABR. In this case eMusics quality will be better. With average bitrate the song you encode *must* be at the bitrate you say it has to have, even if the average bitrate should be higher, while variable bitrate, well, is completely variable. So the 1995 'Foo Fighters' Album in V0 has an average bitrate of ~270 kBit/s, while Joanna Newsom's 'Ys' only has ~206 kBit/s. In the first case there would be bitrate shortages, in the second case there would be wasted space by applying ABR instead.

Better explanation: http://wiki.hydrogenaudio.org/index.php?title=Average_Bitrate

Any time foobar2000 is recommended, popular components should be mentioned as well (also a link to that hydrogenaudio-hosted wiki). My essential components (all on that linked page):

• Columns UI
• LyricsDB
• Monkey's Audio decoder
• Autoplaylist Manager

Also, foobar2k includes transcoding settings for LAME MP3 and Nero AAC, but the binaries aren't included (for licensing reasons I assume). Foobar2k asks for the locations of "lame.exe" or "neroaacenc.exe" the first time you try to encode to these formats. They can be downloaded for free:

• LAME MP3
• Nero AAC

Finally, essential reading for newbie foobar2k users: Bachi-Bouzouk's Guide to Foobar2000 v0.9.

Any time foobar2000 is recommended, popular components should be mentioned as well (also a link to that hydrogenaudio-hosted wiki). My essential components (all on that linked page):

• Columns UI
• LyricsDB
• Monkey's Audio decoder
• Autoplaylist Manager

Also, foobar2k includes transcoding settings for LAME MP3 and Nero AAC, but the binaries aren't included (for licensing reasons I assume). Foobar2k asks for the locations of "lame.exe" or "neroaacenc.exe" the first time you try to encode to these formats. They can be downloaded for free:

• LAME MP3
• Nero AAC

Finally, essential reading for newbie foobar2k users: Bachi-Bouzouk's Guide to Foobar2000 v0.9.

Any time foobar2000 is recommended, popular components should be mentioned as well (also a link to that hydrogenaudio-hosted wiki). My essential components (all on that linked page):

• Columns UI
• LyricsDB
• Monkey's Audio decoder
• Autoplaylist Manager

Also, foobar2k includes transcoding settings for LAME MP3 and Nero AAC, but the binaries aren't included (for licensing reasons I assume). Foobar2k asks for the locations of "lame.exe" or "neroaacenc.exe" the first time you try to encode to these formats. They can be downloaded for free:

• LAME MP3
• Nero AAC

Finally, essential reading for newbie foobar2k users: Bachi-Bouzouk's Guide to Foobar2000 v0.9.

http://wiki.hydrogenaudio.org/index.php?title=Foob ar2000:Increasing_Performance

Damn Right. Fix up metadata and file names without all the crap that the Godfather or Musicbrainz put you through:

http://wiki.hydrogenaudio.org/index.php?title=Foob ar2000:Components_0.9/Masstagger_(foo_masstag)
http://wiki.hydrogenaudio.org/index.php?title=Foob ar2000:Titleformat_Reference#.24caps.28x.29

Damn Right. Fix up metadata and file names without all the crap that the Godfather or Musicbrainz put you through:

http://wiki.hydrogenaudio.org/index.php?title=Foob ar2000:Components_0.9/Masstagger_(foo_masstag)
http://wiki.hydrogenaudio.org/index.php?title=Foob ar2000:Titleformat_Reference#.24caps.28x.29

It's just whining. There have been numerous double-blind ABX tests, many done by the folks over at Hydrogenaudio.org, comparing MP3 files to AIFFs, and with the right codec and right bitrates (depending on the type of source material), it's possible to get an MP3 that only the most refined ears can discriminate from the original. [1]

Of course, it's quite possible to make an MP3 that sounds like a tin-can telephone with one end held underwater, and I'd argue that many of the consumer-ripped files floating around the P2P networks fall into this category, but these files only exist *because* there aren't legitimate, professionally-made, DRM-free MP3s. (And because some people like getting stuff for free and don't much care about the quality when they do. But I do think there is a market for and profit in digitally-delivered music, for the people who can do it right.)

As more music begins to be distributed as MP3s, sound engineers will doubtless (if they have not already) begin studying the codecs and encoding procedures in order to wring the most quality out of a particular bit rate. Many amateurs and enthusiasts have already done this, and there is a sizable body of work devoted to the topic -- including the LAME encoder itself.

Also, looking towards the future, while CDs have pegged the standard for digital music as 2 channel, 44.1kHz, 16-bit PCM, there is no reason why an appropriately-crafted MP3 file cannot *exceed* it in terms of quality. The Apple iPod already supports (slightly) higher sample rates, I believe, and if consumers desire it [2], there's no reason why modern digital formats cannot encapsulate very high-definition audio.

The only people who I hear whining about MP3 are those with either an ulterior motive and a desire to try and keep the industry from moving away from a distribution model that revolves around physical objects, or those who just don't understand the technology. (There are a very small core of audiophiles and techies who seem to dislike MP3 because they prefer some other format, usually either for ethical/political reasons or technical ones, and there certainly is an argument in favor of using lossless formats in lieu of MP3 for distribution, but overall MP3 strikes a good balance between quality and portability. [3])

[1] One 'competition' that pitted serious self-described audiophiles against modern codecs is described in detail here: http://www.geocities.com/altbinariessoundsmusiccla ssical/mp3test.html. While well-trained ears could discriminate between 128kbit MP3s and PCM, they could not reliably tell the difference between 256kbit and PCM, on average. This is just the tip of the iceberg.

[2] Which is a big 'if.' The buying public, to date, has shown little interest in high-definition audio as such. The only exception to this is multichannel audio, but that only in movie soundtracks for surround sound.

[3] This does raise the question, though, of why the legitimate music-download sites don't take a cue from the late, great, AllOfMp3.com and just allow the *customers* to choose their format of choice for their downloads. There's really no particular excuse not to at least offer a few different quality/size options, particularly for popular music that is going to be enjoyed in a variety of settings (automobiles, portables, home stereos -- each lends itself to a slightly different EQ and compression).

No, you completely missed what I was saying; *both* Sound Check AND Replay Gain require you to analyze the audio directly, if the tags do not exist. They function *identically* in that regard, it's just the precise algorithm used when scanning the files and determining the gain which differs. Apple could have supported Replay Gain *exactly* how they support Sound Check, as far as users are concerned (with some additional logic if they wanted to support album mode).

The algorithm is described here. Whatever Sound Check uses doesn't appear to be documented, and it seems to lack the "Audiophile" (Album) version described here. Either way, both involve a one-time analysis of the audio (which can and often is done by encoders like LAME, just as Sound Check is likely done by Apple's encoders, but this is by no means necessary; it just saves work for the player), and the use of tags to store the result, and both *attempt* to do the same sort of thing.

Going by what you're saying about Sound Check, Apple didn't bother with some of the more subtle aspects of determining a suitable gain (that, or album mode makes a real difference; I never really tried Replay Gain in single track "Radio" mode). This makes the comparison with ALAC more apt, really; ALAC's somewhat similar to FLAC in design, but misses some details which result in it having both lower compression ratio and higher decode complexity.

No, you completely missed what I was saying; *both* Sound Check AND Replay Gain require you to analyze the audio directly, if the tags do not exist. They function *identically* in that regard, it's just the precise algorithm used when scanning the files and determining the gain which differs. Apple could have supported Replay Gain *exactly* how they support Sound Check, as far as users are concerned (with some additional logic if they wanted to support album mode).

The algorithm is described here. Whatever Sound Check uses doesn't appear to be documented, and it seems to lack the "Audiophile" (Album) version described here. Either way, both involve a one-time analysis of the audio (which can and often is done by encoders like LAME, just as Sound Check is likely done by Apple's encoders, but this is by no means necessary; it just saves work for the player), and the use of tags to store the result, and both *attempt* to do the same sort of thing.

Going by what you're saying about Sound Check, Apple didn't bother with some of the more subtle aspects of determining a suitable gain (that, or album mode makes a real difference; I never really tried Replay Gain in single track "Radio" mode). This makes the comparison with ALAC more apt, really; ALAC's somewhat similar to FLAC in design, but misses some details which result in it having both lower compression ratio and higher decode complexity.

Hydrogen audio forums provide a lot of very good information including well designed double blind comparisons between codecs and bits rates. See this page for details and the links to other testing sites. http://wiki.hydrogenaudio.org/index.php?title=List ening_Tests All in all an excellent resource for any serious listener.

They are about equivalent in file sizes. Of course, FLAC has the edge in being open source, thus being supported on more platforms with less worry about the format being abandoned.

The iPod probably would have used FLAC instead of ALAC had the iPod been powerful enough to decode it on the fly or had been lighter on the CPU and battery. It seems like a waste to invent another format (R&D, support costs) when a free one works just as well (see Apple's usage of KHTML, etc.).

The rumor is that FLAC may be supported in Mac OS X Leopard by default (and presumably in a forthcoming version of Quicktime) -- now the question is whether iTunes will support FLAC metadata (please iTunes devs, if you are reading, do this!).

Wow. Just wow. I had no idea. The built-in SPDIF in my Macs doesn't do this; neither does the older Creative card in my Win/Lin box. That certainly would #*&@ everything up. If I bought a card and discovered it was sending 48 kHz PCM to my receiver when playing tracks ripped from CD I'd probably think it was set up wrong and try to reconfigure it. Thanks for setting me straight.

Scary, huh? :) Hydrogenaudio has a bit about it, and I also found this page which specifcally mentions Creative as being bad offenders! The biggest things to watch out for it seems are any of the Audigy cards and anything which is labelled AC97. In some cases I've even heard of cards/drivers which take 44.1kHz from the player, then upsample to 48 for the mixer and then back down to 44.1 for output! Wacky...

and that source is... ?

The advantage to ALAC is that it has a nice transport - mp4 (m4a)

can you please describe how m4a is a nicer transport? anyway, FLAC can be encapsulated in m4a the same way ALAC is so it's a moot point.

and nice encoder (iTunes)

no one can add support for a codec to itunes except apple. if you would like FLAC support in itunes please tell them

Performance is neck-and-neck, otherwise.

I ask yet again, please provide some evidence for that claim. it is totally counterintuitive once you understand the design of both codecs, that FLAC's much lower decode complexity by nature will translate more easily into a faster decoder implementation, which is corroborated by the evidence I gave.

Source simplicity, which matters none to real people, is much in ALAC's favor. FLAC looks an awful lot like other Xiph products' source - very busy, and very little whitespace (i=1+23|more; all over the place), and SOOOOO many files, even if it compiles to a rather small 40 KB (decoder only)... ALAC's source, ala Hammertime(ton), is a stroll in the park (easy) compared to FLAC's busy downtown streets and back alleyways (forever lost).

I don't see what that has to do with anything, but your "ALAC source" by Hammerton is based on reverse engineering and only supports a subset of ALAC. a FLAC decoder which only supported such a subset of FLAC would be drastically shorter.

Relatively, no one uses either, but more no ones use FLAC.

please provide some evidence for that claim. lossless is a niche, yes, but among that niche FLAC is much more popular. this is also intuitive since FLAC has been around longer, is supported in many more devices and software, has more features, is non-proprietary, is faster, and compresses more. but in any case here is some evidence:
2007 HA poll
2006 HA poll

and that source is... ?

The advantage to ALAC is that it has a nice transport - mp4 (m4a)

can you please describe how m4a is a nicer transport? anyway, FLAC can be encapsulated in m4a the same way ALAC is so it's a moot point.

and nice encoder (iTunes)

no one can add support for a codec to itunes except apple. if you would like FLAC support in itunes please tell them

Performance is neck-and-neck, otherwise.

I ask yet again, please provide some evidence for that claim. it is totally counterintuitive once you understand the design of both codecs, that FLAC's much lower decode complexity by nature will translate more easily into a faster decoder implementation, which is corroborated by the evidence I gave.

Source simplicity, which matters none to real people, is much in ALAC's favor. FLAC looks an awful lot like other Xiph products' source - very busy, and very little whitespace (i=1+23|more; all over the place), and SOOOOO many files, even if it compiles to a rather small 40 KB (decoder only)... ALAC's source, ala Hammertime(ton), is a stroll in the park (easy) compared to FLAC's busy downtown streets and back alleyways (forever lost).

I don't see what that has to do with anything, but your "ALAC source" by Hammerton is based on reverse engineering and only supports a subset of ALAC. a FLAC decoder which only supported such a subset of FLAC would be drastically shorter.

Relatively, no one uses either, but more no ones use FLAC.

please provide some evidence for that claim. lossless is a niche, yes, but among that niche FLAC is much more popular. this is also intuitive since FLAC has been around longer, is supported in many more devices and software, has more features, is non-proprietary, is faster, and compresses more. but in any case here is some evidence:
2007 HA poll
2006 HA poll

the newer iPods use very aggressive dynamic downclocking of the playback CPU, and cannot handle the sudden bitrate boosts in some LAME files.

>>> "mediamonkey claims to handle 50K+ files without slowing down. It's amazing what you can find in seconds with google =) The search was mp3 media manager."

I'm not the questioner but this is a bugbear of mine.

I haven't used this software. However, when I ask a question about software I'm looking for answers from people who are knowledgeable about the subject. I can google, but just because manufacturers claim it works for 50k files doesn't mean it does ... if you use it and have 40k+ files on it, I'd want to know. I'd even suffer a "no-one has suggested this which I'm planning on buying due to the mad props it's getting at www.generic-media-managers-web-forum.com.

On google you found a potential solution. But I think it takes a little longer to find a real answer for a real life situation ... mediamonkey does look promising however:

[quote: http://www.hydrogenaudio.org/forums/lofiversion/in dex.php/t42649.html%5D
djiezes
Mar 27 2006, 06:27
Well, on my lousy system (amd duron, 256mb ram, ...), MediaMonkey is pretty slow and crashes when doing some operation on too many files. And compared to foobar it is slow.
It is however easier and more straightforward to use than foobar, which would be an advantage for less tech-savvy people.
Big_Berny
Mar 27 2006, 06:40
Interesting because I only have an Thunderbird 1.2GHz and 256 MB Ram too! biggrin.gif And here it doesn't crash, at least the newest version. But you're right Foobar is faster. (Although MediaMonkey is also way faster than winamp...)

Big_Berny
[end quote]

Further searching suggests that some people have stability problems and that Mediamonkey have good astroturfers (and some happy and not-so happy users) and bad mp3 codec usage. Generally handles large libraries (reports are generally for about 2-4k files) well.

fb2k is known for being very effecient, even in the face of crazily huge libraries. I dare say you'll hate the default interface/config, but it's not difficult to bend it to your will (though it's not exactly iTunes; more like vim/mutt for music).

Windows only unfortunately, though it is supposed to work well in Wine. Significant chunks of it are BSD licensed.

fb2k is known for being very effecient, even in the face of crazily huge libraries. I dare say you'll hate the default interface/config, but it's not difficult to bend it to your will (though it's not exactly iTunes; more like vim/mutt for music).

Windows only unfortunately, though it is supposed to work well in Wine. Significant chunks of it are BSD licensed.

fb2k is known for being very effecient, even in the face of crazily huge libraries. I dare say you'll hate the default interface/config, but it's not difficult to bend it to your will (though it's not exactly iTunes; more like vim/mutt for music).

Windows only unfortunately, though it is supposed to work well in Wine. Significant chunks of it are BSD licensed.

fb2k is known for being very effecient, even in the face of crazily huge libraries. I dare say you'll hate the default interface/config, but it's not difficult to bend it to your will (though it's not exactly iTunes; more like vim/mutt for music).

Windows only unfortunately, though it is supposed to work well in Wine. Significant chunks of it are BSD licensed.

fb2k is known for being very effecient, even in the face of crazily huge libraries. I dare say you'll hate the default interface/config, but it's not difficult to bend it to your will (though it's not exactly iTunes; more like vim/mutt for music).

Windows only unfortunately, though it is supposed to work well in Wine. Significant chunks of it are BSD licensed.

fb2k is known for being very effecient, even in the face of crazily huge libraries. I dare say you'll hate the default interface/config, but it's not difficult to bend it to your will (though it's not exactly iTunes; more like vim/mutt for music).

Windows only unfortunately, though it is supposed to work well in Wine. Significant chunks of it are BSD licensed.

Forgot to mention that you'll probably want to go to hydrogenaudio instead of /. for these sorts of questions.

I was going to post something to that effect, but you beat me to it. So I will add to what you just said the following.
- as soon as Vorbis gains popularity, you can expect every patent troll out there to search for conflicting patents, so Vorbis may not be any better than another proprietary format in that regard. At the very least, choosing a proprietary format gives you someone to sue if the format you chose is infringing.
- furthermore, Vorbis is not such a hot choice for portables right now because of poor battery life.The playback time when playing Vorbis may be reduced by as much as 33% as compared to mp3 playback.

Not that I'm a proponent of the much-improved WMP 11 on the Windows platform (I prefer foobar2000). But I do like AAC now that Nero has made freely available (for Windows) a very high-quality VBR AAC encoder that's supposedly better than iTunes/Quicktime (which makes very good CBR AAC files). Sure, it's not as "free" as LAME, but Windows users no longer need to install iTunes to encode AAC files.

AoTuV: http://wiki.hydrogenaudio.org/index.php?title=AoTu V

Hehe, what I love is that the ppl clamoring for higher quality tracks are ppl who couldn't pass a double blind test between probably 96-128 kbps and 192. The truth of the matter is that double blind tests are rarely conducted by humans above 128kbps because it is so difficult(I've heard of a high bitrate test using some program to look for noise). In fact there is a 48 kbps test(Thats right, Forty Eight Kilo Bee Pee Ess ;) test in progress at the respected http://www.hydrogenaudio.org/forums/index.php I was looking for another thread about how "ABX killed my ego" but gave up. Basically, no one can tell the diff easily above 128 using modern encoders, its all the placebo effect. Download foobar2000 http://www.foobar2000.org/ and try taking a double blind(ABX test) with FLAC vs 128 kbps lame. Have fun flipping a coin to decide which one is higher/lower quality.

Using up to date encoders, for the vast majority of people, for the vast majority of tracks, 128 kbps is indistinguishable from source.

Link.

Everyone should try to ABX at least once. You'll be shocked how much worse your ears are that you'd believe them to be... ABX Just Destroyed My Ego is a very informative read for any would be audiophiles:

I think the reason is in large due to the common misconception that audio compression heavily alters the sound. Less dynamics, weaker bass and all those other descriptions "audiophiles" like to throw around, and that in fact are nothing more than just placebo. But in reality, the artifacts are much more subtle, and often require actual training for an inexperienced user to be able to hear them.

Using up to date encoders, for the vast majority of people, for the vast majority of tracks, 128 kbps is indistinguishable from source.

Link.

Everyone should try to ABX at least once. You'll be shocked how much worse your ears are that you'd believe them to be... ABX Just Destroyed My Ego is a very informative read for any would be audiophiles:

I think the reason is in large due to the common misconception that audio compression heavily alters the sound. Less dynamics, weaker bass and all those other descriptions "audiophiles" like to throw around, and that in fact are nothing more than just placebo. But in reality, the artifacts are much more subtle, and often require actual training for an inexperienced user to be able to hear them.

Using up to date encoders, for the vast majority of people, for the vast majority of tracks, 128 kbps is indistinguishable from source.

Link.

Everyone should try to ABX at least once. You'll be shocked how much worse your ears are that you'd believe them to be... ABX Just Destroyed My Ego is a very informative read for any would be audiophiles:

I think the reason is in large due to the common misconception that audio compression heavily alters the sound. Less dynamics, weaker bass and all those other descriptions "audiophiles" like to throw around, and that in fact are nothing more than just placebo. But in reality, the artifacts are much more subtle, and often require actual training for an inexperienced user to be able to hear them.

Yes, there will be a quality diff between #4 and #1, but it'll be the same miniscule PSNR loss as from #1 to #2. So unless you transcode a dozen times or something it won't really hurt you.

Every encoder will generate ringing and other artifacts. Every good encoder tries to put those artifacts just a bit below the hearing threshold according to an algorithm that has been tested extensively with normal music. However, encoders are generally not fine-tuned to deal with the unnatural type of noise that results from another encoding process, resulting in the noise ending up above the hearing threshold after the second time.

You might wish to check some double-blind test results on HydrogenAudio. Short version: reencoding 256 kbps MP3 to 128 kbps MP3 sounds horrible compared to 128 kbps MP3 straight from the lossless source.

Carma-whore. "Lossless is without quality loss" - What an obvious carma whore! Burn this idiot, please!

You could at least have provided a link to Wikipedia or the HydrogenAudio wiki.

Please notice how THIS post is posted as AC, yet not anonymously.

Cheers,
KingOfGod

Actually, with the iPod I would consider that annoyance to be an equalizer that for some reason or another results in digital clipping (a.k.a. distortion) when you use it regardless of volume. Nevermind the fact that the EQ presets are preloaded by Apple and that there's no way to change those or add your own.

Foobar 2000

It's FOSS, so the GUI is generally crap (it's as unitiuitive as other media players while still being ugly and unskinnable by default) but it's very lightweight and unobtrusive.

Here's what it looks like with one person's customizations and skins: http://wiki.hydrogenaudio.org/index.php?title=Imag e:Columnsui.png

Since Column UI is so popular, I think its site needs to be much easier to access from Foobar2000's front page. Also, to add a "volume slider" (not enabled by default - WTF?), right-click the toolbar area in Columns UI and add the volume slider.

Foobar 2000

It's FOSS, so the GUI is generally crap (it's as unitiuitive as other media players while still being ugly and unskinnable by default) but it's very lightweight and unobtrusive.

Here's what it looks like with one person's customizations and skins: http://wiki.hydrogenaudio.org/index.php?title=Imag e:Columnsui.png

Since Column UI is so popular, I think its site needs to be much easier to access from Foobar2000's front page. Also, to add a "volume slider" (not enabled by default - WTF?), right-click the toolbar area in Columns UI and add the volume slider.

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

http://www.hydrogenaudio.org/forums/index.php?show topic=48231&hl=

Then go to Hydrogenaudio and check out/join the [discussions on] listening tests.

Re: "Point A": I wouldn't necessarily make that assumption -- in fact, if evidence like this is to be believed, VBR AAC will in some, if not most, cases be larger than CBR at the same bitrate settings. Speculation that Apple's "VBR" is in fact a variant of the comparatively-inefficient class of ABR algorithms is also worth noting, though I can't speak to its veracity.

>mp3 is the standard and although it has its flaws isn't all that bad.

The same thing can be said about Internet Explorer vs Other Browsers. "Bohoo, IE isn't all that bad. Installing alternative browsers is a pain (notice I said pain, not impossible). Bohooo, life is so hard, I might have to plan one step ahead, push one more mouse button, bohoo..". Man, that's just fucking lazy. Nobody said having a sense of quality and a little fire for what's right would be the easiest way to glide through life.

Feels pretty darn good to not be a punter of the mp3 licensing consortium though. YMMV.

And honestly. FLAC and APE get about 2:1 compression. So does ZIP, GZ and BZ2 when applied to PCM. What, exactly, is the difference?

No, they don't.

The difference is:

APE: %55 of original size

FLAC: %59 of original size

RAR: %67.5 of original size

BZip2: %84 of original size

GZip: %91.5 of original size

There's a reason why new audio-specific lossless compression algorithms were developed. Back in the day, the performance of other general compression algorithms was even worse with audio - RAR has seen a lot of improvement, for example. But even with improvements, they can't come close to audio-specific compression schemes. Also, audio-specific compression schemes are aimed at LOW requirements for decompression.

It just so happens that FLAC and APE are two of the earliest formats to show significant gains over RAR, so they're firmly seated in the top spot now.

Foobar2000: http://www.foobar2000.org/ It works well with my ~2000 MP3/OGG/M4A and Mod tracks. It's fast, has a small footprint, is expandable, has library features, it's extremely customizable, works nearly out of the box with most standard multimedia keyboards, has a decent following and is actively developed. It's also GPL. I could babble on about the plugins I use but I'll let the HydrogenAudio wiki do my talking for me :) http://wiki.hydrogenaudio.org/index.php?title=Foob ar2000:Components You'll be most interested in the ColumnsUI stuff but there's quite a few other gems in there. HTH

Yup. Linked right from the official ColumnsUI site. There's also a lot of stuff on the forums, like this thread. Azrael is my current favourite.

Yup. Linked right from the official ColumnsUI site. There's also a lot of stuff on the forums, like this thread. Azrael is my current favourite.

foobar2000 has a tabbed interface with separate playlists in each tab which is nice. I like the sparse interface. Some people hate it, although if you are willing to invest the time there are a lot of ways to customize it to make it look much nicer.

The Columns UI is enabled by selecting the "Foobar2000" menu, then selecting "Preferences," then "Display," then changing "User interface module" from "Default User Interface" to "Columns UI." I think it should be easier to find the Columns UI, but I don't want to complain too much about a great app with so many great customization options.

Here's an example of what Columns UI can look with a few more customizations:

http://wiki.hydrogenaudio.org/index.php?title=Imag e:Columnsui.png