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Audio Compression Primer
Hack Jandy writes "For those of you with a little extra time this afternoon, check out Sudhian's primer to all things concerning audio compression. The article details everything from DRM to CRC matrixes (with a healthy dosage of Ogg)."
"FLAC is the Linux users lossless audio codec of choice"
Unless your doing some form of audio editing or "production" recording, is lossless really worth the extra size compared to a 192kbps Ogg or MP3? I usually have more problems with static from the stupid 3.5mm jack than a lossy format.
I'm sure "SlashdotMedia" will improve on all the wonders that Dice Holdings blessed us all with
Each to their own, but I am more than satisfied with oggs or mp3s encoded at a reasonable bitrate - I think the popularity of hardware such as iPods suggest that most other people are too.
What was that trash? "FLAC coding is kind of like run length encoding."
Yeah, kind of, except that you'd be lucky to get a sub-unity compression ratio using RLE on sampled music... FLAC != pkzip for crying out loud, and even pkzip is out of the league of this "primer".
That and phrases like "your compressed file might use up more data" are nauseating. How do you use up data? I think he meant "disk space".
I know that even large radio stations use 128Kbit sampling frequency. I have heard musicians saying they cannot distinguish the difference between the audio sound played by CD and MP3 with 128Kbit encoding. I have switched from 128K to VBR 320K but just because "that is a good style".
MySQL Error 1040: Can't return sig, Too many connections!
Not very informative for slashdot ppl. I think we should have had an article more about code or something. I think most slashdotters understand codecs and the differences in lossless and lossy compressions. Waste of 15 minutes.
Shouldn't that be 1200 kb/s? 150 KB/s * 8 = 1200 kb/s, right? Or is the 150 KB/s figure I'm using incorrect (I could have sworn that was the 1x CD speed)?
I did a little googling and found this (http://www.teamcombooks.com/mp3handbook/13.htm):
Because the code is open source, FLAC will be around forever and available on whatever OS/Platform you want to use it on if you feel like compiling the software.
Another reason it's going to be around and much more prevalent as time goes on is that the compression is so good and the speed/resource usage figures are so attractive. When I rip CD's to FLAC I am limited to 40x by my burner (CPU utilization is around 20-25%). When I rip the same CD to ogg, I top out under 30X because the processor has reached 100% utilization.
Fast. Free. Efficient. Frugal with the CPU. What else do you need?
Ok, mod me down if I'm clueless, but on the first page: "Compact Discs use a bit depth of 16, allowing for 2 ^ 16 possible levels."
I always thought CDs were encoded in 12 bit, not 16?
https://www.accountkiller.com/removal-requested
Call me crazy, but I insist that there are certain 'killer' tracks where I can hear this distortion even at higher bitrates in advanced MDCT codecs like Vorbis, namely Led Zeppelin / Rock and Roll whose drumline consists of a ridiculous number of cymbal crashes in rapid succession.
The way I see it, the future is lossless. With hard drives burgeoning to over 500GB and Fiber-to-the-Home becoming a reality within the near future, why bother saving a little extra space at the cost of degraded quality, which, the more you listen to audio compressed with a certain transform, the more likely you are to hear distortions? I think in the future we'll see a greater trend towards lossless audio compression with codes like FLAC and its ilk.
Page 3:
These are all mathematically lossless codecs. They theoretically should sound identical when using the same hardware and playback software. I don't believe I could conduct double blind or any other subjective test which would prove beyond any doubt that there are any differences between the options aurally.
-> When something is mathematically lossless, proving the encoding / decoding algorithmes is the only way to go. Double Blind, Subjective??? I'm really sorry I read the whole thing, I want a refund for my time!
While the article is a primer, I was a little disappointed in the algorithmic treatment given in the article itself. Right now I know of two excellent free publications: Introduction to Sound Processing and The Sounding Object, which both treat the theoretical, DSP side of things. Any other resources that Slashdot readers can recommend for those who are interested in the subject of audio compression and representation?
Titus Barik
I know that everyone says that transcoding loses quality. So when I finally got an iPod, I initially transcoded all of my .ogg files to .mp3, but planned on re-encoding later from the CDs. But when I re-encoded a few tracks with lame, and listened to them versus the transcoded track, I honestly couldn't tell a difference (even with headphones). If I ever notice poor quality on my iPod, I guess I'll re-encode then, but otherwise why bother?
I believe that I used the default settings for both oggenc and lame, which was ~128 bit VBR in both cases.
especially when listening to music on hi-quality speakers a la Bose.
bose?? high quality? you must be out of your mind.
This article is already outdated. :) His discussion of lossless formats excludes the iTunes Apple Lossless encoder, which is supported by iPod. From their websites Import Music page:
Weapon of Choice
"However, you can choose to use different audio formats for any track that you import from CD. iTunes lets you convert your music to MP3s at high bit-rate for no additional charge. Using AAC or MP3, you can store more than 100 songs in the same amount of space as a single CD. Discerning customers and audiophiles want true CD audio, and now iTunes can give you that quality with the new Apple Lossless encoder. You'll get the full quality of uncompressed CD audio using about half the storage space. You can copy music in this format onto your iPod or iPod mini, to take perfect audio wherever you go"
https://www.accountkiller.com/removal-requested
"For those of you with a little extra time this afternoon..."
;P
Congratulations, you just defined a Slashdotter.
"Mind, as manifested by the capacity to make choices, is to some extent present in every electron." -Freeman Dyson
I have a question about OGG ever being supported in a player. OGG is under continuous development to make the compression better which is why they have the rating system of 1 through 5. This would mean that a 2 today would probably not be the same size as a 2 a year from now.
My question is would a Digital player be able to play all variations of the OGG even while it is upgrading? If not then without firmware upgrades through the life of the model it wouldn't work. Otherwise if you wanted to make OGG files a year from now they wouldn't work on your player.
If I'm wrong about this please let me know.
background noise? at 193kb? you are the one who needs you get a decent stereo. most likely a crappy DAC. or maybe pot.
and if you are familiar with the musician's business, you'd know that almost all of them do appalling things to their ears. just go to any concert and you'll know the damage that musicians do to themselves. and that simply goes with the business. a business you obviously aren't very familiar with.
I second that.
3 ,pg,1,00.asp.
:
On repeated double-blind tests on very expensive equipment, even audiophiles are unable to distinguish between CD quality and LAME encoded 192 kbps MP3 files. Those who say they are able to aren't using double-blind tests or have super-human mutant ears. If you go check over at Hydrogen-Audio (where audiophiles and people who care far too much about LAME settings hang out), most of the forum posts indicate that anything above 192 kbps is transparent even to their equipment, which is pretty above average.
On regular equipment, PC World did a small test a while ago on standard equipment: http://www.pcworld.com/reviews/article/0,aid,6412
Their results found that ~192 kbps is pretty much transparent as well.
mp3-tech.org also has a listening test availible. On their run, they found 192 CBR kbps to be nearly transparent (*feels* different, but don't know why), and 256 kbps CBR to be completely transparent (can't tell compressed from source CD).
"The listening equipment is the following
* Teac VRDS 25 CD reader
* MIT T2 cables
* Yamaha AX 1050 amplifier
* Denon PMA 960 amplifier (for frequencies 50Hz)
* Celestion speakers"
This test was also done a while ago on an older mp3 compression program( c. 1998), so current LAME encoding probably allows for complete transparency at 192kbps or so.
Some guy has a law that says you need to sample at a rate twice as frequent as the signal your sampling. Makes sense if you think about it.
That would be Mr. Nyquist. In practice, you get about 80% of the ideal bandwidth due to a non-zero transition width in the anti-alias filter and extreme group-delay at passband edge.
To be precise, you have to sample at twice the bandwith of your signal. For a lowpass signal (audio would count), this is twice the highest frequency present. For a bandpass signal (eg, RF), you can sample at twice the bandwidth of the signal(*) even though the actual frequency is much higher. This technique is known as under-sampling.
(*) Assuming the input bandwith of the sample-and-hold circuit on A/D is sufficient.
(S(SKK)(SKK))(S(SKK)(SKK))
Vorbis decoder is and has been done for a long time. Like other codecs, tweaks can always be made to the encoder to produce better results by using different psychoacoustic models, etc. As long as the output still follows spec, the decoder will still decode just fine. This is why your crappy MP3's from 1997 still play today, and fancy MP3's from today will still play on those old sound players from 1997. As long as the encoder follows spec, the decoder will always be able to decode it properly.
Any one else having a problem reading that page with Opera? I get this white rectangle which obscures part of the text.
No, matrices.
I hate grammar Nazi's.
They're so into compression they compressed their text so much you can't read it! :)
Seriously, their text is 10 pixels high. That's too small to read on anything but a very low resolution monitor. Even with my new 23" LCD wide-screen, 10 pixel high text is unreadable. Hey, how about learning how em's work before attempting to publish on the internet? There are way too many idiots now that claim to be webmasters that don't have a clue.
(Mod to -3, nitpicking)
The MDCT in itself is actually lossless. Any distortion you notice is most likely introduced by the quantization applied post MDCT during compression.
"There is no dark side of the moon really. Matter of fact it's all dark."
I know, I know, a "primer" Useless article....
I use CDs. No problems yet.
Will report with more later.
Letter
WRONG!
Nyquist's criterion is "You must have at least twice as many samples as the largest BANDWIDTH of the signal in order to correctly reconstruct it."
You can take a 10.7 MHz signal, and sample it at 10000 samples per second, and correctly reconstruct it, so long as the signal is guaranteed to be bandwidth limited to 10.7 MHz +/- 2.5 kHz. This is often done in software defined radio to aquire the signal from the intermediate frequency (IF) of the analog front end.
You also have to have an appropriate reconstruction filter at the output of the system in order to correctly recover the signal - if you don't have the right reconstruction filter, you will NOT reconstruct the signal correctly.
You also have to take into account the effects of any signal modulation - take a 20 kHz sine wave, and burst it for 10 msec, and you widen the bandwidth of the signal by about 100 Hz (depending upon the exact shape of the burst - a perfect square burst will widen the signal as a sinc function and will, in effect, increase the bandwidth to infinity, which is why square bursts are generally Considered Harmful in communications work).
Also, you don't oversample a signal in time to account for "rounding errors" - you oversample in time because the frequency response of sampling a system in time introduces a sinc response in frequency - by moving the sampling rate up you reduce the impact of this response on the recovered signal's frequency response. You also greately ease the requirements on the reconstruction filter - the filter can be wider (have fewer poles in the transfer function - thus fewer parts needed).
www.eFax.com are spammers
Um, no. 20/20K is more accurate, and we lose a kHz every 5-10 years as we get older.
Most of the time I am content with a good Ogg encode (I mean, hell, I'd never have heard the difference if the samples weren't played back to back!) I generally only use FLAC for a) my favorite albums and b) classical music. Size wouldn't be an issue... but for the fact that I keep an oft-updated mirror of the data on a second computer. As drive space is become rather inexpensive, I forsee a time when lossless will be the way to go, except for portables.
*Ascend Acoustics CBM-300 stereo pair, HSU sub, and a HK AVR-325 receiver.
A preposition is a terrible thing to end a sentence with.
(As an Engineer who has thoroughly studied ADC/DAC) I would say that the article presents a very good background on the issues of sampling and reconstruction of audio.
However, the rest of the article is approached from the heavily biased opinion point of an "audiophile", which the majority of the population is not. These audio experts have fantastic equipment and a keen sense of hearing, allowing them to distinguish between the subtle difference between high fidelity recording and playback. Such people like software like foobar2000 and care a lot about dynamic range, and for the most part think that lossy encoding is a shame. This is a bit about being picky, and a bit about showing off, but either way it's a minority viewpoint.
But such people are by far the minority of the public. Most of us don't get caught up in the subtle details of audio recording and playback, partially because we don't care, and partially because we don't have the fine equipment (electronics and human ear) to notice such things. So the article for instance completely dismisses lossy encoding, even though this is by far the most exciting frontier of modern audio compression. You can get 64 kbps (ogg vorbis) or 32 kbps (aac) streams that sound amazing to most people, as good as FM radio.
As an Engineer that is what I find exciting, because we can transport "essentially the same" amount of media in far, far less bandwidth than it required a decade ago. And the efficiency is improving all the time, ditto for video.
I had a Vorbis listening party this past Summer at my home. I told everyone to bring their favorite CDs and that I would rip their favorite track from the disc using MP3 and Vorbis. I did so at 64, 128, 256 and 384 k bitrates. We had a wonderful time conducting blind listening comparisons using both the AKG and Sennheiser cans as well as my Tannoy studio monitors, Yamaha stereo speakers and the Bose 901 series loudspeakers. (Each set in a room seasoned with the best in acoustics) Under such discriminating environments, Vorbis beat MP3 hadns down every time. Some people couldn't even tell the diffrerence between the 64k Vorbis and the 256k MP3. ONly going to prove my point that Ogg Vorbis is FAR superior to any other codec.
After we had dinner (a fine French meal with wine if you must know), it was time for more listening tests. Initially the crowd was a little resistant, but by the time we'd listened to the wonderfully executed "Get Ur Freak On" by Missy Elliot for a fifth time, the crowd agreed that Ogg Vorbis was the winner hands down.
It was a wonderful day and a great victory for Ogg Vorbis as I told everyone present that now that they were aware of the quality the Vorbis provides, they should show all of their friends and family. I provided them all with archive DLTs of the test set (music that they all brought with them) so that they could give it to their IT guys at work and have them load it up on the their Linux or Unix servers and share with their colleagues. They all promised that they would talk to their IT guys.
So, this article has no idea what it's going on about because it isn't aware of the change that is taking the nation by storm with regard to Ogg Vorbis. I urge all of you to show your family and friends the right way to archive digital audio media and advise them to abandon MP3 and proprietary codecs. If you don't then it will be on your own hands...
- Audio formats supported: AAC (16 to 320 Kbps), MP3 (32 to 320 Kbps), MP3 VBR, Audible, AIFF, Apple Lossless and WAV
- Upgradable firmware enables support for future audio formats
The second bullet leaving the possibility there, but the page lists it as currently (meaning iPod users now, popularity etc) not supporting it.For context, click Parent.
Sampling frequency would typically be 44.1KHz, bitrate would be 128kbps. Also, FM radio quality (with good reception) compares to about 96kbps well-encoded mp3, so there's not much point in them recording higher except for archival purposes.
You should be using LAME to encode, and LAME only goes up to 320kbps (blade for instance goes up to 384kbps, but is much lower quality), ergo you can only have 320kbps CBR, not VBR.
And to everybody else out there who complains about background noise, you should be extracting digitally from the CD!
flac doesn't seem to have come far enough yet for me (500+ albums is a lot of diskspace if it's around 300MB/album), but to my ears on my equipment (Klipsch £250 (pound sterling if that doesn't come out) speakers, cheapo SB Audigy2 soundcard), lame --preset standard (around 200kbps VBR) sounds damn near perceptual transparency.
You know you've been IMing too long when you almost say 'lol' out loud to a non-geeky friend...
I had a Vorbis listening party this past Summer at my home.
But no one came.
While Ogg Vorbis encoders and decoders are still in development for the purposes of tweaking the compression and bugfixing, but for all intents and purposes, the bit stream format has been set in stone back with the 1.0 release quite a while ago.
All encoders should be compatible with all decoders (with the exception of some extreme encoders such as the 2Kbps encoder). Vorbis is no longer a moving target (like Theora still is at the moment) and if you make a decoder you can be assured that it will play all Vorbis files from the past (since 1.0 of the format), present, and future.
Xiph.org couldn't have made it any easier for hardware manufacturers, providing the integer codec Tremor (to run on embedded processors) and don't charge royalties for using or modifying a Vorbis codec in a hardware player (almost all other formats are patent-encumbered and charge royalties).
Nature of material. The CD on my computer desk is Pete Tong's "Twisted Beats" - a DJ mix in which bass predominates and vocals are highly processed already. Although I haven't tried, I'm sure I could listen to that material for hours at a highly-compressed ratio without hearing any problems. The CDs stacked by the stereo in my living room include, e.g., Bach violin concertos recorded by Itzhak Perlman. Because of the numerous high transients in such material, I'm sure they'd stink in mp3 of anything less than the highest bitrate, and maybe even that. I've not tried this particular CD, but my experience encoding classical music in mp3 at anything lower than the highest bitrate has been dismal, and resulted in my throwing the files away. (In addition to violin, soprano vocals and all pieces recorded in a acoustically-rich environment such as a large church particularly stink as mp3. I've not tried them as OGG but I assume the same would hold. I also suspect the same is true for heavy-metal or other rock music that uses guitar distortion algorithms rich with high harmonics.) So perhaps when arguing about whether a particular bitrate or codec is transparent, a reference to the type of material would be in order.
Individual hearing variation. I had my hearing tested at a job once, where they were worried about people losing their hearing because of high ambient noise and failure (not me) to wear ear protection around machines. The tech repeated the test at the highest range about 3 times, as I recall, and said that "I'm just re-doing it because we rarely have anyone who can hear that accurately up in that high range." The highest testing range where he found me unusually acute was only 10-20 kHz. We always read that the range of human hearing is 20 Hz to 20 kHz. Rarely is it acknowledged that this is an average, and that individual hearing will vary, especially at the theoretical extremes of human hearing. I therefore have no beef with people who can't hear high transient distortion, or with people who can't hear such highs at all (and thus are mystified by my pain when I listen to certain poorly-mixed CDs which fail to de-emphasize the extreme high range). But I do get annoyed at accusations of snobbery - I can hear it, OK?
No, no, no. This is not a sig.
"Lossy compression" has a very specific definitition, which is different from the definition of "digital sampling".
OK. so if I take a 16 bit 44.1 kHz stereo waveform, mix/downsample/quantize it into 8 bit 22.05 kHz mono (8:1 data reduction), and turn it back into 16/44 stereo, then we're using "digital sampling" as a crude form of lossy compression, right?
Um, that noise is from the soundcard/computer.
Xiph.org couldn't have made it any easier for hardware manufacturers, providing the integer codec Tremor (to run on embedded processors)
There are two kinds of devices that can theoretically decode MP3 but not Vorbis. Vorbis, even with the Tremor fixed-point implementation, generally requires more arithmetic operations and memory accesses per sample than MP3 does, and a device that decodes MP3 in real time at 95% CPU utilization (using the other 5% to drive the user interface and the storage device) may not run the Tremor decoder in real time. Worse yet, some players have the MP3 decoder on an ASIC that takes an MPEG audio bitstream on one pin and produces PCM on another, with no way to reprogram it for Vorbis.
Many other posters have discussed the scant coverage of lossy encoding.
There is a distinction between Ogg and Vorbis that is lost in the summary (and much of the discussion). Ogg is a container format which can hold many other kinds of data (video like Theora, audio like Vorbis, and lyrics in a format which is being worked on, just to name a few) including combinations of data encoded with various codecs. So the lossy encoding in question is Vorbis, not Ogg.
Just because a program can understand the container does not mean that a user running that program can play the encoded performance. One should recognize this distinction so one can begin to understand what is going on with Ogg FLAC and Ogg Speex files. These files are not common, but we're better off understanding how things interrelate, even in a broad sense, not just memorizing a bunch of filename dot extensions.
Digital Citizen
Just put on a pair of Sennheiser HD580 or HD600 headphones, and you will EASILY hear the difference between 192kbps MP3 and uncompressed audio. And I do mean, easily. Even people who don't know what to listen for hear the difference and run to the store to buy HD580's. :0)
Also of interest might be this article over at Ars Technica: "A guide to ripping and encoding music". It discusses some of the more practical aspects of music ripping and encoding.
What about musepack? It seems like this codec is constantly passed up; yet in my own testing, and double-blind testing with friends and family, they chose the MPC files 80% of the time over OGG and MP3.
Also, if you do a frequency analysis of the raw input compared to MPC's --standard setting output, there's very little difference, where as MP3/etc. will do a "round" or "drop off" after a certain frequency, usually 16-20kHz.
Anyways, hydrogenaudio.org is a great site for information about all this stuff..
Not All Who Wander Are Lost
I've got about 350GB of lossless audio goodness in a set of nice oak bookshelves built into my wall. Considering that the time it takes to get up, get a CD, rip it, and encode it is not much longer than it takes to locate a FLACed album on my fileserver and encode it - that is, the encoding stage is several times longer than the "get up and rip the first track before starting to encode" phase - I think I'll stick with my current system.
Dewey, what part of this looks like authorities should be involved?
http://cs.fit.edu/~mmahoney/compression/
http://www.msoftware.co.nz/WinRK_benchmarks.php
http://cs.fit.edu/~mmahoney/compression/
http://www.msoftware.co.nz/WinRK_benchmarks.php
The one advantage to having them on your computer already in lossless format is that you can encode multiple CDs faster than you would be able to sitting there and putting each CD in one after another.
yeah like removal of harmonics from your fft means its totally loss less?????? wtf
Soundproofing Acoustics noise
the man that had that idea did not realise what harmonic removal does at a psychological level.
Soundproofing Acoustics noise
When I read your comment about Led Zeppelin/Rock and Roll, I thought you were full of crap, but thought I'd test first before commenting.
So, I skipped to that song in my playlist, and started listening, and at first it all seemed ok. Then, just before the first "Lonely Lonely Lonely Time" (about 50 seconds into the song), I heard it. (This is Vorbis with quality level of 7 - default is a mere 3)
You bastard. I would never have noticed that if it weren't for you.
mp3 does damage your aural cognition
Soundproofing Acoustics noise
Toss the crappy little earbuds, stop buying 99 cent songs that sound like crap, learn to really HEAR the music. Those $15 speakers on your PC are worthless. You should go buy decent speakers (they cost all of $40), but if you insist that the free speakers that came with you PC are 'good', then maybe you're hopeless.
Rip at 192 or better. That's sort of settled in as the minimum. A few years ago, 128 was the 'miminum' because it didn't sound shitty but it could be 'shared' easily. But that's been replaced. Like 5 years ago. Catch up. The download sites like Napster, iTunes, Whatever, are serving up the 'good stuff' as it was declared to be seven or eight years ago! That's crap. Their 128 kbs junk isn't worth 9 cents, let alone 99 cents.
Make your peace with it: Either you don't care that you're not hearing all the music, so 128 kbs is fine, or, you can and do hear the difference, and you'll not d/l (and definitely not pay for) anything less.
Off soapbox now. :)
Sig not available, please try again later. If the problem persists, then the submitter is an idiot.
The poster's point is valid. So's mine. I say. Shun the crap below 192kbs :)
Sig not available, please try again later. If the problem persists, then the submitter is an idiot.
>>So, people who would benefit from a 24/96 ogg are extremely small in number.
h tm
Have a read: "There's Life Above 20kHz!"
http://www.cco.caltech.edu/~boyk/spectra/spectra.
Feel free to jump to section X & XI (results).
I haven't been able to find too much more, newer work done on the subject, so I don't think there is a great deal of scientific interest in it. Interesting read though.
A preposition is a terrible thing to end a sentence with.
Flac is very much like run length encoding, in fact run length (ie DC signal) is one of block types.
FLAC says take this starting point and extend it for X samples by applying one of four very simple formulae to generate each subsequent sample. Then take that signal and add this residual signal to it (which has a very small amplitude so can be expressed in a small number of bits) and you've got the output signal.
It is WAY WAY simpler than a lossy codec, and thinking of it as having the RLE concept at its core is a perfectly reasonable way to understand it.
If you only compared Ogg to MP3, how can you claim "my point that Ogg Vorbis is FAR superior to any other codec."
it's also a polemic against lossy compression which is not appreciated.
I'd go on a Vegan diet but the delivery time from Vega is too long. --brownkitty
Simple. MP3 is far superior to any other codec except Ogg Vorbis. Why do you think everyone and his brother uses MP3? I'll give you a hint: It's not because it sucks. The MP3 Pro codec that came out later was really improvements upon the original MP3 codec that were bolted on from the Ogg Vorbis project. WMA is a joke. No one would ever consider WMA a serious codec.
ba-dam, tchza!!
I use FLAC for full quality archival, stored on my computer on the *other* 160GB HD. The way HD prices are dropping (less than $80 for 160GB; less than $200 for 300GB on NewEgg.com as I compose this response) pretty much anyone should be able to fit their entire collection on a large secondary drive (unless you have an exceedingly large CD library or are rather enthusiastic about offering "backup services" to all of your friends for their CD collections).
When I want to go mobile with my music, I transcode to VBR mp3 since I can pack more music/GB that way; give the portable music player industry a few years and we'll have iPods packing upwards of 200GB and then we'll just dump our FLAC library to the iPod and goove out in lossless mode.
It's easy to test it yourself (if you have windows). winabx is a program for performing ABX-style listening tests. ie. You can change the sample "on the fly" and you should identify if the music played is sample A or sample B. After some runs it will tell you if you really identify the samples or if you are just guessing. I found out that with the genre I listen to vorbis encoded with -q4 is enough for me. Earlier I used -q7, which I know now to be overkill for me.
Considering that the time it takes to get up, get a CD, rip it, and encode it is not much longer than it takes to locate a FLACed album on my fileserver and encode it
Do that for five albums. You have to stay by your computer and change CDs at intervals. Now do it for fifty albums. Now do it for five hundred albums.
When you are talking about one or two albums, fair enough, there isn't much difference. But your solution doesn't scale well at all.
The hardware you use to listen is composed of two pieces of equipment: Your speakers and your ears.
See this other post, and before you start asking, I encoded with lame, with the r3imx archive CBR profile (at least for the 256kbps track). And this is the second time I do this kind of test with two different people. So there is obviously a difference for him between 256kbps and uncompressed.
And remember that if average joe cannot tell the difference with his $200 speakers, he will be disappointed when he'll buy (in 20 years) some nice audio setup and realizes he can throw away all of his MP3s.
Write boring code, not shiny code!
But, what happens when your original cd collection (or even one disc) is lost, destroyed, or stolen? You're left with absolutely nothing.
With a proper (lossless) archive, you can recreate your entire cd collection, bit for bit, if you had to.
Others have mentioned speed of encoding to other formats, and that's another great advantage. I have scripts to convert my entire flac collection (over 200 discs) to mp3 in one swoop.
Does anyone here think that this guy is off his rocker? An "Ogg Vorbis Listening Party"? While I think that Ogg Vorbis really is one of the best codecs for lossy audio compression, I find this story hard to believe. Insightful? Please. I mean, for god's sakes!!! He gave all of his party goers a data tape to share with the network admins? What admin in his right mind would mount a tape from a non-IT person who said their brother-in-law or uncle gave it to them to propagate the Ogg Vorbis listening samples? If I had an admin who did that on my network, he'd be fired in an instant. We're talking IP theft here if this guy is telling the truth. Wake up people!!! This HAS to be a troll! Pull your heads out of your colletive ass and moderate appropriately.