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Non-MP3 Codecs?
Vanth Dreadstar asks: "While
MP3 is okay, I have begun researching other codecs that would be
suitable for my home music use. Lossy codecs such as Ogg
Vorbis, AAC,
and MPC all seem to have promise, not to mention the lossless codecs
such as Shorten
(otherwise known as .SHN),
LPAC, and FLAC.
I would like to know what non-MP3 codecs people are using out there,
and why."
Ogg Vorbis over MP3 because obviously Ogg is free while MP3 is locked up in patents, and if you're one of the golden-ears that can tell the difference, FLAC for high quality (and still free).
I'm using .nap because Napster is going to come back! Just you wait!
Whatever it is that comes on these shiny round things I get from the music store...that's the one I use.
"On the Internet, nobody knows you're a dog!" - a dog
i use a non-lossy format known as the Audio Interchange File Format, or AIFF, to store my audio files. They can be burned to CDs very easily -- you can't fit as many on one CD as MP3, but the CDs will play in every CD player I've come across, and the sound is CD-quality.
go get it
I use Ogg becuase:
1. it seems to give better sound quality for the same quantity of bytes.
2. encoding to Ogg is legal, unlike encoding to MP3 when using ISO-code based encoder (pretty much any encoder i know. enlighten me if im wrong).
3. "Ogg" sounds cooler than "MP3"
I'm using Ogg Vorbis for a number of reason. The reference encoder, while not perfect, is certainly not bad. The vast majority of the time, .ogg's sound noticeably better than MP3's of the same bitrate.
.ogg files with the track names grabbed from FreeDB. To actually encode, one symply drags the .ogg file to another directory, and the IO slave works its magic.
More importantly, Ogg Vorbis is free of any patents or any other restrictions. I could make a commercial hardware player if I wanted to, and not have to pay any royalties to anyone.
Finally, it integrates nicely with Konqueror's audioCD IO slave. You can simply type "audiocd:/ogg/" in Konq's location bar, and it shows you a list of
Lex orandi, lex credendi.
Another consideration is the straightforwardness of the API for the library you intend to use. Vorbis has a somewhat reasonble API with a liberal addition of quirks. Also you can easily add metadata to Vorbis files. Ever tried adding metadata to an MP3 file? ID3v1.1 is trivial but ID3v2 has a 95,000 line reference implementation. Uh? UH?
Any application has to support PCM audio also, since most music collections are primarily on CD.
i dont think hes going for smaller files, but better sound, therfore why not a format like AIFF or even... .wav, it may make huge files, but who cares?
ZAP (an acronym for "Zero-loss Audio Packer") is, as its name implies, lossless, and the ZAP app has the ability to play back audio from a compressed archive.
The ZAP application compresses raw audio files to about 40-to-70% of their original size. This is much better smaller than typical .zip or .sit compression on audio files.
Archives can be made self-extracting. I find this useful if I do an audio project for which the files total about a gig in size but want to back it up to a single CDR.
Interestingly, I just looked at emagic's web site, and they do not have a link for ZAP. Maybe their site is incomplete, or maybe they have discontinued the product.
I'd like to know if they are using them.
"If you think education is expensive, try ignorance" - Derek Bok
Of a good old wav. Except maybe for pure vinal, but is that a codec?
"as plurdled gabbleblotchits on a lurgid bee" - Prostetnic Vogon Jeltz. (One man's humorous is another mans flamebait)
is the *best* lossy audio codec I've yet seen. At -q 3 (ends up being around 112 kb/s average) most is transparent to me, and at -q 4.99 pretty much everything. (I don't use -q 5 because it jumps up to lossless coupling which makes the bitrate jump quite a bit).
Aside from sounding great, it's 100% free (open source, patent-free) for everyone, and I can always annoy people on #vorbis (opn IRC network) with technical questions.
If you're looking for lossless compression, wait for the people currently working on vorbis to write Ogg Squish, which will be their lossless codec, and should kick ass as well.
I'm also looking anxiously forward to Ogg Tarkin, the currently-in-the-works lossy video codec, which is using new technology (wavelets) to encode video. I believe it shows a lot of promise.
--
grep "xercist"
I recently bought a Rio Volt MP3/WMA CD player, and compared WMA8 with VBR LAME, and LAME won hands down. Both encoders are set to come out around 128kbit, and while both of course have artifacts, the artifacts in WMA are MUCH more noticeable. I guess I'm just alot more sensitive to the type of artifacts WMA produces...
Because I have to quit this filthy .mp3 habit. I need the music industry to help me overcome my addiction to free music, so with digital content controls I won't be tempted to download gigabyte upon gigabyte of free music. I won't have to continue working this extra part-time job to support my purchases of extra hard drive space.
My sig hates me. That's ok, I never cared for it much anyway.
Grip is a nice front end to Linux command line ripping and encoding utilities. You can choose which encoder you use and I believe it already has a preset configuration for ogg encoders.
Sticking with MP3s is a no brainer unless you have to use open software for moral reasons, since Apple has enhanced MP3 encoding/decoding for AltiVec, and this is an area where those gigaFlops do wonders at quick, high-quality encodes and freeing up more CPU for your work (or the visualizer :) during playback.
"Reality is just a convenient measure of complexity" -Alvy Ray Smith
is a ftp database and crawler similar to audiogalaxy only for ogg. it would catch on in no time.
For most distributed applications (music player in my living room) I use the MP3 side. If push came to shove, I'd find some way to delete the MP3's and play the WMF's on other devices, just because they're so space-conscious.
Because the few songs I have ripped are in that format, and the few songs I get from friends now and then, are also mp3.
:)
I don't really play "clog the modem", so I guess I am the wrong person to answer that.
But I am not going to play the elitist game of switching to Ogg because it has better compression (cheap HD, cheap bandwidth) or because it preserves some frequencies more (come on, you can't hear it either).
I could think to switch just because of the licensing and the patent issues, I am like that sometimes... but right now it is too much trouble to make a point noone will notice (as I share my music as much as I DC for new - almost never).
I do personally hope that for those that this really matters to, that something like Ogg will come and take over, so we can see AOL buy that too. Just kidding.
It still doesn't matter to me. If I could listen to WMA on my linux system(s) I would. If I could use WMA on my car mp3-cd player, I would.
I can't though, so it doesn't matter. I'm not a musician by any means, nor can I detect the difference between 160 and 192 mp3 compression. So I'll continue using my inferior, yet cross platform, non-license restricted, used-everwhere, mp3 format.
Subject says all. Has it just not hit mainstream, or is it getting steamrollered by Ogg, WMA, and any of the other popular formats?
In the end they will lay their freedom at our feet and say to us, Make us your slaves, but feed us. - Fyodor Dostoyevsky
Bear in mind that the ~4x compression rate listed for lossless compression schemes is heavily reliant on the input. Don't be surprised if you get 1.5-2.5 compression a lot of the time, and remember that there's a good chance you'll get 1:1 (or worse) compression results with a 'random' enough song file.
Obliteracy: Words with explosions
JPEG users have available to them some command line utilities that permit simple alteration of images without loss of quality, for example, rotation and flipping. Are there any similar utilities available for any of the major audio compression formats?
The reason I ask is that I have ripped a number of CDs and the volume levels vary noticibly. I like to listen to MP3s as I work, with the volume turned down far enough that I can hear the music, but any one that I'm on the phone with won't. Unfortuately, there doesn't seem to be a single setting for everything that I've ripped. While I could go back and re-rip, I'd much rather have a toolbox of useful batch utilities. Ideally, it would allow me to write, say, a Perl script that generates a histogram, checks the average and peak volume, and then tweaks a single number in the file header to force it in line with the rest of my collection.
Is this sort of thing possible?
Nothing for 6-digit uids?
I can't really tell much of a difference between 128kbps mp3 and the original cd. Maybe others can, but mp3 is plenty good enough for me. As is ogg. To me, it doesn't really matter about the format as long as its convienent. And considering the 200+ cd's of mp3's are full of mp3's and no other format... and the effort required to convert them would outweigh the slight gain by converting to another format.
-Restil
Play with my webcams and lights here
Try grip. Configure it to rip to any format you want - all it needs is the path to the executable. It will do the freedb lookup and name the files in your favorite style too.
Since ogg vorbis is a free codec (as in beer; as in speech) this is really the best way to go. Note that US Linux users who rip to MP3 with free-as-in-beer software are probably in violation of one or more laws. Since XMMS plays OGG as well as MP3 you can mix and match MP3's from your favorite P2P community with OGG's of your own collection.
as you probably know there is a sparse few number of them available to download...
So what are you waiting for? Get oggenc and do your part!
-Renard
I wrote one that can encode using multiple mp3 codecs, or can use oggenc.
it is command line, but I at least feel it is quite easy to use, and it is faster then alot of them, because as soon as it finishes ripping a song, it forks an encoder, then goes on to rip the next.
anyway: it's called the One Ripper, and it is at:
http://www.evilsoft.org/Software.
it requires a linux system with perl 5, and it has links to some perl libraries you need.
Enjoy.
- The unexamined life is not worth leading -
I like midi. But I've heard on this new thing called mod, it takes samples and tone shifts them to recreate the song! Pretty cool, like midi but better!
in the mean time - I can't stand mp3s, ogg might be the way for me to go now.
* Much better than OGG and MP3
* Picture perfect at 128 kbit/s
And what is this comment based on? These results have been pointed out in comments for previous articles, but I'd like to mention them again. ff123 has been conducting double blind tests comparing various audio codecs, and the results are here.
The following is from the page:
Comparisons in red below are true as a group with 95% confidence.
ogg is better than wma8
mpc is better than wma8
ogg is better than xing
mpc is better than xing
aac is better than wma8
aac is better than xing
lame is better than wma8
lame is better than xing
Looks to me like WMA8 got beat by pretty much everything... But hey, what good is statistical analysis anyways...
besides the fact that it's hard to go up against an established standard and the fact that there is no hardware support, is that storage is so cheap now. If I can get a 60GB drive for under $100, why would I want to sacrifice a big chunk of processing power to make my music 1/3 smaller? Only if I absolutely wanted to use something open.
"Reality is just a convenient measure of complexity" -Alvy Ray Smith
It's free, as in beer.
You can stream it! And a little app called abcde works great with it.
It's slowly becoming a new standard are more software players are supporting it.
Too bad there is no hardware support. I think we should start off with a DC port. What do ya' think?
Get your Unix fortune now!
Editing with 1-sample resolution, for example. This allows you to cut your live music into tracks without that silly gap introduced by mp3.
Support for 256 channels, channel coupling, etc, are also extremely important for streaming applications.
...however, I don't see a format shift happening too soon, since the majority of computer users (the "dumb masses", I like to call them) are being spoon-fed by the OEMs, and we all know what they're using in place of strained peas. Not even Winamp support is enough; nowadays, every Compaq/HP/Dell/eMachines/craputer is pre-configured with Windows Media Player or RealOne, and they don't support OGG or the others (mostly because no one can profit from them).
"Ancillary does not mean you get to rule the world." --U.S. Circuit Judge Harry Edwards, speaking to the FCC's lawyer
I'm not interested in some "super small" music file - disk space is cheap and MP3 is already small enough for transfering over the Internet. I'm more interested in audio quality and hardware compatibility. MP3 and WMA sound great (moreso the latter), and are both commonly supported by cool hardware. I don't see the point in all these other media formats. I like to listen to my music on something other then my computer.
There is no longer anything that can be done with computers that is nontrivial and clearly legal. -- Paul Phillips
I will probably continue to use mp3 format files, because it is basically the standard that everyone on the internet goes by. If you have broadband and a decent hard drive, size/bitrate should not be a deciding factor. Unless you're one of those that hear like a dog, mp3s should be sufficient for everyone to use.
In most cases, a 60kbps OGG file sounds as good as an 128k mp3. An 80k OGG is as good as 160k mp3 and half the size.
Actually, Ogg only shaves off 30-40% (still respectable, just not revolutionary)
If you have a portable player, you would appreciate the smaller size with high quality.
If you have a portable player, you almost certainly can't use Ogg's :)
If you make computer games, you have a high quality free way of adding a lot of music to your games. (possibly patents for mp3)
If you want background music in a computer game, why would you want to use a format that eats drastically more processing power?
You can do 44.1khz and 48 khz audio.
So can MP3, what's your point?
The encoder sounds good by default, so music traded on file sharing systems sounds good (unlike all those terrible 128k mp3s encoded by anything that isn't LAME).
So "The Encoder" for MP3 is bad? If there was just one encoder this would be an argument. And I do hat those 128k bastards just as much as you :) At least iTunes defaults to 160k.
Now the other points are very valid, but they probably won't get anyone to switch at this point. What we need is a format that gives at least 4x the compression of MP3 with the same quality (and reasonable CPU usage) to get people to switch. Hopefully it will be an open technology like Ogg.
"Reality is just a convenient measure of complexity" -Alvy Ray Smith
I teach Computer Science at the high school level at a largish school near Austin, Texas. For the past several years there's been a "jukebox" in my room where students could vote for albums to hear during programming lab time, and random tracks off the winning albums play over the speakers in the classroom.
Over Christmas break I changed the "player" portion of the system to play Ogg Vorbis files instead of mp3s.
Why not mp3?
So, then, why Ogg Vorbis?
By the way, if you haven't listened to Ogg since 1.0-rc3 came out (on New Year's Day), try it again. The sound quality has been much improved. Note that you should not use the "-b" option to encode as it uses CBR and thus produces larger files at lower quality. Default is quality 3, which is 112 kbps but sounds as good as 160 kbps to most. If you really can tell the difference, quality 4 averages 128 kbps and sounds much better (and is maybe 3% smaller) than an mp3 at that rate. You've got to experiment to find your own sweet spot.
The biggest downside is that whole ubiquity thing. There's been an official Winamp plug-in for quite some time, but Nullsoft have yet to install it by default (rumor has it that it is AOL 's legal department which is holding this up). I'm also pretty sure there's a Windows Media Player codec, but don't quote me on that.
Also the only hardware player that supports Ogg Vorbis is the HipZip (via a firmware upgrade). Other units that support it are coming soon, but not yet available.
Since I don't own a hardware player (yet) and don't download my mp3s, the ubiquity factor isn't an issue for me, however.
On the plate for rc4 is sound quality tuning for the low (a.k.a streaming) bitrates. Then a coat of polish and it'll be called 1.0
Graham "Teach" Mitchell, computer science teacher, Leander HS
Microsoft cheats with WMA8 - all they do is compress the range a little which results in an average 3 dB volume level boost. It has been repeatedly shown in multiple independent studies that even just a small increase in volume makes music "sound better" to the average listener. Often you'll get all kinds of superlatives about accuracy, openness, full-bodied, etc, etc from the people comparing the louder track to the quieter one. MS knows this which is why they play those psychoacoustical games with WMA8.
If you compare a good mp3 encoding (say with LAME and the right arguments) to a WMA8 encoding of the same bit-rate and with the volume levels matched, mp3 will win out, or at least tie, everytime and Ogg will usually do the same with 25% less bits.
I use MP3s because they're much like Interet currency.
:)
I convert MP3s to WMAs when I want to squish music onto my PocketPC.
If I bought an OGG car player (if there is/was such a beast), I'd convert my MP3s.
The point: When in Rome, I do as the Romans. It's a simple life, really.
WhatEVA
There's a batch Ogg replaygain tool at: http://sjeng.org/ftp/vorbis/
ReplayGain tself is explained at: http://www.replaygain.org
The latest XMMS plugin already supports replaygain (as does latest Ogg123), and it should be in the Winamp plugin soon if not already. Right now it's up to individual apps to support ReplayGain, but we're deciding on an easier way to encourage/include support with core Ogg.
Monty
Thats a deceptive statement. WMA at 128k may sound better than MP3 at the same bitrate, but there is no way to encode WMA at quality comparable to high-quality 256k/320k MP3 files, or the LAME VBR settings which can produce almost-indistinguishable files at ~200k average bitrate. I suspect the same may be true of ogg, but at least its Free so if it can't do high bitrates now it may be modified to do so in the future.
And none of the lossless formats are even as compact as 320k MP3. So MP3 still fills a useful niche in that regard.
(yes, I'm aware that since LAME isn't licensed, its technically illegal, but my, and I suspect your, primary use of it is pirating copyrighted music, so its not as though using a patent-free codec would make what we do any more legal.)
"(Man) tries to live his own life as if he were telling a story. But you have to choose: live or tell." --Sartre
1) CDex. Has an Ogg encoder (RC2 version) embedded, and you can use the command line RC3 version with it very easily. The latest betas use the 'cdparanoia' libraries to rip. This would be nice choice once it's been updated to RC3.
2) EAC. This is the benchmark for quality ripping in Windows. It's slightly harder to set up, and doesn't integrate as nicely with passing metadata to the external ogg encoder, but it's the best Windows ripper bar none. Both pieces of software are free. CDex is also open source (useful if you happen to have a copy of VC++ floating around).
-- Help Digitise the Public Domain at DP.
I'm not familiar with the state of MP3 tools which support ReplayGain, but I know that Gian-Carlo Pascutto just wrote a tool to add ReplayGain information to Ogg Vorbis files. There is an XMMS support in CVS which uses the information, and I just got done adding support for ReplayGain to ogg123 (it will be about a week before it goes into the xiph.org CVS pending the approval of some other changes). Winamp also supports ReplayGain using Peter's Vorbis plugin
Ogg has had high bitrate from the beginning. It will happily take you up to just under what the lossless codecs will give. in rc3 stereo, -q10 will do ~400-600kbps, and -q0 will give you ~48-80kbps depending on material.
Monty
I can hear the difference between a 128kbps mp3 and the original CD (192kbps CBR or 160kbps VBR are good enough for me), however the difference isn't nearly so great as the difference between playing the music on $30 vs. $100 speakers. You can get decent computer speakers today (if you're not an audiophile and don't need very high volume) for as little as $60, but the prevalence of 128kbps recordings on the internet suggests to me that most of these people are still listening to music on the little white buzzers that came with their computer.
Another one to try is Normalize It alows you to adjust volumes across different types of input files (.wav, mp3, etc...)
Grip is just about as easy as it gets. It comes with Red Hat 7.2 now, preconfigured for ogg, preconfigured to query the freedb servers for tracks and titles. I'm still using it with lame, but when I move everything to ogg it makes that easy, too, with "Auto-rip on insert" and "Auto-eject when finished" boxes checked.
what you suffer from is lack of normalization. many many CD's are poorly mastered (in fact 90% of all Cd's today are very poorly mastered, it is very rare that anyone takes the time to properly master a CD anymore.) what you are getting is that the mixdown mastering was set at an arbitrary level by the studio staff. they just picked a level and spun off a master without running a calibration on the equipment. They usually calibrate every morning, but many places assume that the calibration was good from yeaterday, and the equipment wasn't touched or turned off so just fire away.... they have 300 albums to master today... this usually leaves you with CD's that have a horrible noise floor because the audio program is too low and not using the entire abilities of the CD. (NOTE there are some that are messed up the other direction.... Nutral-milk-hotel comes to mind.. clipping on the cd because it was not normalized.)
so you need to normalize up. basically use a program that looks at the entire song and then brings the higest peak up to 99% or 98% of max. the program will look at either each track, or all tracks from an album, find the highest peak from that album and then normalize all to that peak. either eay works great, I prefer each song getting normalized.
Now... you can do this to mp3's you have already. problem is that you need to decode-normalize-reencode which adds more loss and noise artifacts.
I would start over, grab your cd collection and start from step one again. (lame has awesome encode now... it's improved massively)
Do not look at laser with remaining good eye.
This is not exactly right. To keep in line of the rest of your examples it would have to look like this:
4. Musician with Internet only distro:
MP3 -> uncompressed format -> MP3 -> uncompressed format -> MP3
2 generations of lossy copying
MP3 is definitely a lossy encoding method in that every time it is decoded there is a good chance that you will not get out EXACTLY what you encoded in the first place. You will instead get something that sounds close enough that the human ear can effectively treat them as the same. The problem is that artifacts tend to crop up with each encoding and you will most likely end up with garbage after a few encoding/decoding cycles.
You are correct in that you don't need to encode/decode and then encode again to copy, however that is true of your options 2 and 3 also. Once your data is in digital form you never need to encode it again, just do a lossless digital copy and it is likely that you will never lose quality. This has nothing to do with codecs, but rather with the nature of digital data.
Sapere aude!
.WAV *is* PCM. With headers that differ from the PCM files on audio CD's (.CDA). As has been pointed out elsewhere, PCM is simply a way to describe audio data using ones and zeroes. There's no compression involved.
News and bla for computer musicians: http://lomechanik.net/
2. Original CD -> Tape -> Tape -> Tape 3 generations of lossy copying.
3. Original JPEG -> save as JPEG -> save as JPEG
2 generations of lossy image manipulation.
Hence the term lossy
While that is an interesting way of looking at it, you are the one misusing the term "lossy".
When it comes to compression, lossy has a specific meaning - it means you can NOT recreate the original input bit-for-bit. With lossless compression, you CAN recreate the original input bit for bit. It has nothing to do with percieved quality.
In the future, please make sure you know what you are talking about before accusing others of ignorance. :)
Check out DRM-free movies at http://www.bside.com
Imagine if all porn sites would store their pictures in .gif format (or even better .bmp) and all Napster users would use .wav.
The wasted bandwidth caused by Code Red would be insignificant by comparison...
P.S. Ogg is the way to go
* Picture perfect at 128 kbit/s
I don't know about the rest of you, but to my ears, NOTHING is "picture perfect" at 128kbps. 192 is minimum for any lossy compression.
Very simple:
Step 1: Insert CD
Step 2a: Type "audiocd:/ogg" into Konqueror's URL bar and save it as a bookmark.
or Step 2b: If you already have saved a bookmark, get it.
Step 3: Drag the .ogg files to your music folder, they will be compressed on the fly.
Can't become easier than that.
Undeniably true. But established standards die enventually. MP3 R&D has been mostly abandoned. It will be around for a very long time yet, but it's being attacked from all technological sides. Microsoft wants to kill it for WMA, Tompson wants to kill it in favor of MP3 pro, FhG wants to kill it for AAC, Real wants us to use Real--ermm, sorry, ATRAC3, etc. MP3's been superceeded and abandoned by cutting edge research.
MP3 the king is a mighty warrior, but he's showing new wounds. Ogg is the successor to the throne, and the only codec individuals are going to have ready, unrestricted access to once MP3 eventually falls. It's not happening this year, but it's happening.
and the fact that there is no hardware support
A mostly fair thing to point out. Ask again in a year; the FPU-less codec exists (he says, hacking on ARM7 assembly), now it's mostly the business distribution arrangement that's up in the air. Commodity hardware designs can't quite live in the same open framework as software.
is that storage is so cheap now
Most of the big Geek music collections of friends around me are each over a Terabyte of music. That's still alot of money.
If I can get a 60GB drive for under $100
If quality is not a concern, you can get a cheap turntable for much less than that and it never runs out of space.
why would I want to sacrifice a big chunk of processing power to make my music 1/3 smaller? Only if I absolutely wanted to use something open.
This one confuses me slightly...
Compressing from WAV->Ogg makes things ~10-20x smaller, depending on your quality tastes.
If you mean 'why would I replace my mp3 collection I already have?', in that case I agree with you. An equivalent Ogg will sound better/more consistent and be smaller, but if you're satisfied with what you've got, there's no need to replace it. Certainly don't transcode it! It could only end up sounding worse (see rant here)
If you mean, "why would I encode to Ogg rather than MP3; it's not worth it", then you're just confused. You get smaller, better sounding files for no extra effort (and no extra CPU). In this case, Open Source is not a compromise; Vorbis is the best out there. All we're lacking is the portable players.
Monty
That's because if you master it well on the first try, you only get to sell the CD to the fans once.
As copyright owner of this comment, I authorize everyone to defeat any technological measure which limits access to it.
If you're referring to MP3Pro, I doubt it'll ever be used by anything outside of the streaming audio market.
I'll grant that an MP3Pro at 64kbps sounds better than an MP3 at 64kbps, but for purposes of archiving audio for quality (as opposed to streaming), the diskspace savings isn't enough to justify (a) not gaining freedom from Fraun's patents (Ogg wins here), and (b) losing the freedom that comes with a DRM-free codec like MP3 or OGG.
But if you're willing to put up with DRM in exchange for better sound at streaming rates, might as well go with Windows Media .WMA instead of MP3Pro.
I can't imagine anyone on /. who'd be willing to put up with a DRM-crippled codec in the presence of .ogg (if patent-freedom and low-bitrate quality matters) or .mp3 (for availability, archival quality at high bitrates, and a willingness to turn a blind eye to the patent issue).
The more I talk to (to be acurate the more I am talked at by) audiophiles the more I get the feeling that its a geek weenie measuring contest and has nothing to do with what stuff sounds like. One guy I know told me at great length how his $2,000 cd player was superior to the cheapie Philips unit it shares its main circuit board with because of the accuracy and freedom from wow/flutter of the CD drive mechanism.
So when I hear about golden ears and such I tend to think Bovine Excrement.
I would much prefer to use Ogg or Windows Media Player than MP3 because they are better compression formats and allow more tracks to fit on my Archos device. Problem is that the Archos won't play them to the better compression is moot.
I am not that much interested in the Napster/Gnutella scene any more than I am aware of any other WareZ scene so use of the codec by others is not that interesting to me. However if someone came along with a 6Gb Hard Drive of 'stuff' I could well imagine preferring to do swapsies than encoding the stuff myself. Ripping off tracks one at a time over Napster while being spamvertised is not my idea of fun.
Looking for an Information Security student project suggestion?
Try http://dotcrimeManifesto.com/
The reason the live music trading community (most notably etree.org) uses the shorten format is because there was not a way to widely distribute exact copies of, say, master DATs. Now, assuming the person transferring the DAT, did a reasonably good job, every person after that who receives the SHN files can create an exact copy of that DAT. This is crucial because of the way shows are distributed. One person gets a copy from his friend, and he passes it on to his friends. If there was a lossy step involved in the middle of the chain, each copy would be worse than the one before. Note tape trading. Copying a cassette is lossy, so someone who got such with a 4th or 5th generation tape was stuck with all of the artifacts that were introduced in each generation above. Even copying CD audio is not perfect: programs that do digital audio extraction need to do a good job reading the data without any error correction. Shorten makes 100% sure that every copy is just like the original.
>
>Like So?
>1
> 0
> [8 times per sample]
No, it's even worse!
(1-2 k of headers and track metadata deleted)
<BYTE>
<BIT>1</BIT>
<BIT>0</BIT>
<BIT>0</BIT>
<BIT>1</BIT>
<BIT>1</BIT>
<BIT>1</BIT>
<BIT>0</BIT>
<BIT>0</BIT>
</BYTE>
... because it's an alternative.
Not only that, but it's smaller for the same quality output than mp3 or wma.
I do not want to encode my music to something that will cost me in the long term because of OS restrictions. Not long now and Microsoft will force you to buy their mp3 and wma playing licence software. (As has happened with the Windows Media Encoder in XP...)
My 2 cents.
Heh if i had mod points i would totally bump this up. MS paid musicians to say WMA sounded better than mp3 when wma first came out.
:)
I remember hearing that one musician messed up his line and accidentally said "This MP3 track sounds much better than the WMA track" instead of the intended line.
-- Patience is a virtue, but impatience is an art.
Use shorten. With plugins which will allow for realtime decompression and playback (with searching within each song) available for XMMS, Winamp, and Macamp the only issue remaining is storage capacity and processing time involved in decompressing the files. Any Celeron or higher will handle the processing necessary and with 120gig drives well below $300 and 160gig starting to come out...that's a healthy sized cd collection.
A number of online communities use shorten for trading live recordings...www.etree.org is one such organization. WAVs are generated from a number of different sources, compressed, checksum's are generated, then the files are distributed freely.
Another great advantage of shorten is that if something comes along that provides better (or more desireable) compression you can un-shorten all of your files to their original state and recompress them using this newer compression scheme....something that no MP3 (or any other compression scheme that I know of) will do.
It's useful to note that any production WMA decoder/encoder is either Microsoft's code, or if otherwise, must pass Microsoft certification. I.e. even in those rare cases that someone outside of MS gets to mess with that code/format, MS makes sure the result is vetted before it may be deployed in a product. Likewise, all products deploying WMA (e.g. digital audio players) must undergo certification independently of whatever WMA code is used in them. This helps to ensure interoperability and sound quality for ports and embedded implementations.
MP3 on the other hand, is something of a free-for-all w.r.t. the available decoders., and no one (esp. not Fraunhofer or Thompson) has a certification process to validate the quality of the generated bitstream. (c.f. another poster's comments about the merits of VBR LAME vs. WMA).
Somehow I suspect you don't know what bloat is...
since then most of [MP3 encoding] happens on cirrus logic processors or TI DSPs.
However, the TI DSPs that handle floating-point arithmetic are much more expensive. Nobody (except Iomega, and even that's not officially released) has made a portable Ogg decoder because the Vorbis reference decoder from xiph.org uses extensive floating-point rather than fixed-point arithmetic.
If you write a Free integer decoder (or fund writing one), they will come.
Will I retire or break 10K?
Wrong! You are completely confusing the analog vs. digital distinction of generational copy degradation with a specific property (lossy vs. lossless) of a digital compression algorithm.
A compression format is lossy IFF the output from an encode/decode cycle may not be identical to the input. Period. I.e. Playing back your DVcam tape doesn't produce the exact digital data that was originally output by the cam's CCD, due to the lossy compression used in storing that data to tape. This has no bearing on generational loss, which digital formats (uncompressed or no, lossy or no) don't suffer from.
Hrmmm... Actually, as part of my senior project, I did a listening test of some different codecs. WMA8/128kbps, Ogg (RC2) 128kbps, lame mp3 VBR nom. 128kbps. Additionally, I tossed in mp3/256kbps and ogg/256kbps, plus the original source wav.
:)
The subjects were allowed to listen to the reference wav at any time, and otherwise, only knew they were listening to "a variety of encoding schemes." They were asked to rate the sample on a scale of 1-10 vs the original and to comment on why they rated the way they did.
The results: WMA came in dead last. mp3 & ogg at 128kbps were evenly matched, with ogg edging out mp3 by a few tenths. The highest rated samples were the mp3 and ogg at 256, although the ogg won by a significant lead - many times it was mistaken for the wave file.
Here's the interesting bit. When broken into age groups, the majority of the testers (college students, 18-24 years old) were dead on the averages above. The other significant group in the study, people 35 and older, often *did* rate the WMA files as better than the mp3 and ogg. But then again, the range of scores they assigned to all of the samples was much tighter, and they reported hearing far fewer discrepencies between the files. Conclusion: young ears hear better. But then again, I'd hope you'd expect that.
For those wondering, the samples used were taken from Peal Jam's Daughter, Fool's Garden's Lemon Tree, and John William's Duel of Fates, for their wide variety encoding nightmares
A preposition is a terrible thing to end a sentence with.
Ok, i'm just going to assume (based on this post by you and previous posts) that you work at microsoft.
One thing you should know is that i make embedded digital audio players for a living. I have been doing this for years. I have personally worked with every codec except mp3pro, and i doubt mp3pro will ever mature to market viability. i have seen and ported the wma decoder source, in addition to a variety of other minor things i could mention to provide credibility here.
* Much better than OGG and MP3
This is quantitative; most listening tests i have read about state that high quality mp3 encoders (such as lame) and the ogg reference encoder produce better quality output than WMA or AAC. I would guess that this trend will continue; Microsoft makes fast, low quality encoders for their desktop applications so as to provide an enhanced initial user experience. This is evidenced with how WMP behaves - it encodes as fast as possible, but generates low quality (notable artifacts) output, even at bitrates of 96kbps and 128kbps. This definitely refutes the claim that WMAv7 64kbps sounds "as good" as MP3 128kbps.
* Picture perfect at 128 kbit/s
No offense, but are you in the marketing department at MS? My response has to be "I'll believe it when i see it." I dont have the golden ears, but i can still tell 128kbps from cd audio, and i dont see this as changing.
* Supported by hardware (unlike ogg)
This is a flat out lie. Microsoft has ported their WMA decoder to various embedded architectures, but has no actual hardware support. The support is all in software, running on embedded processors. As was mentioned in previous posts, Ogg has been ported to embedded devices just like WMA; it's just a matter of time before it's ported to all devices.
* Next version (Corona) will sport 5.1 Dolby, 24 bit samples, 96khz sampling rate, better compression.
That's nice, except most consumer audio hardware handles 16 bit 44kHz audio, which is what CD audio is. So supporting 96kHz audio might look great on paper, but it does absolutely nothing for you in reality. In terms of 5.1 Dolby, AAC supports multiple channels and look where it's gone - nowhere. Maybe you guys should focus on the features that actually matter?
* Existing hardware will update firmware to support Corona
For the love of jesus. Let me drop you a clue:
* Existing hardware will update firmware to support OGG Vorbis 1.0
Your blind faith in WMAv8 has converted me - i am now a true believer in alternate technologies. I will devote all my spare time to the proliferation of disruptive technology.
Thank you for your support.
Try the below script, you'll need Python, mpg123 and sox, all of which are easy to obtain for Linux. This process stores the volume in the comment as text, you might want to consider storing it at the end of the comment in binary if you use the comment field for real information. There are a multitude of other improvements that could be made to this script (command-line options would be a good start.)
/dev/null 2>&1' % (tmpfile, song))
I also have a fairly simple random MP3 player script that also uses mpg123 with the volume settings generated. It normalizes on a song-by-song basis (unlike many of the player plugins that normalize continuously, making the quiet parts of songs no longer play quiet.) It would be fairly easy to modify it to do album-by-album normalization if you so desired. (Assuming your MP3 collection is well organized.)
#!/usr/bin/env python
# standard Py libs
import os, sys, stat, random
# available from: http://id3-py.sourceforge.net/
import ID3
def compute_volume(song):
tmpfile = '/tmp/randplay%d.wav' % os.getpid()
os.system('mpg123 -w %s "%s" >
p = os.popen('sox "%s" -e stat -v 2>&1' % tmpfile)
v = float(p.readline())
p.close()
os.system('rm %s' % tmpfile)
return v
def recurse(directory, callback):
for i in os.listdir(directory):
path = '%s/%s' % (directory, i)
m = os.stat(path)[stat.ST_MODE]
if stat.S_ISDIR(m): recurse(path, callback)
if stat.S_ISREG(m): callback(path)
def do(song):
if song[-4:] != '.mp3': return song
i = ID3.ID3(song)
v = 0.0
if i.comment and i.comment[0] in '0123456789':
v = float(i.comment)
#v = 0.0 # uncomment this to have the script (re)compute the volume of every file
if v >= 1.0:
print '%s: %f' % (song, v)
else:
print '%s: ' % song,
sys.stdout.flush()
v = compute_volume(song)
print '%f' % v
i.comment = '%f' % v
i.write()
return song
recurse(sys.argv[1], do)
It gets 99.99% compression. I think it's termed "lossy" compression.
Outdoor digital photography, mostly in New Engl
Comment removed based on user account deletion
As long as the original format is 16bit 44.1khz, debating what will give us the best sound quality isn't very interesting, since even the original sounds terrible.
I long for the days when SACD or DVDAudio will give us the joy of listening to music back. The fuckers who stole that from us, simply to reduce manufacturing and shipping costs, should in my opinion be @#$%@$%#6
Rubbish.
The raw score was slightly higher in one of the three samples, but by 0.03 - and the more detailed analysis below shows that, for that sample, *there were NO statistically significant results*. There was nothing even *close* to statistical significance. I realise understanding more than 1 number may be a strain, but it's necessary.
Now, for the second sample there were only two significant results (AAC and Ogg being better than Xing) - and those would only have been significant if you had set up the whole experiment specifically to test whether AAC and Ogg were better than Xing. Given that we are asking a general question, we need stronger statistics -- and as a result for our general question there were no statistically significant results for this sample either.
The third sample is the only one with general significant results -- very strong ones at that. And they say that on this sample, Ogg, MPC and AAC were better than WMA8 (LAME may be as well, but the result is a little off the required significance level).
So, given these samples, and these listeners, we can only conclude that WMA8 is certainly not the best codec at 128kpbs. This doesn't imply anything about performance at other bitrates, of course. (WMA8 is probably still the best at 64kpbs, for example).
-- Help Digitise the Public Domain at DP.
May I point out that they don't need to mess with the player to make everyone switch over to the encrypted only files..
In fact, I believe that they make WiMP only encode encrypted files with DRM built in. I seem to recall an associate of mine complaining about ripping all his music into wma with WiMP. Then he lost a specific file, and had to rip all his music over again.. Even though, it was still taking up space on his hard drive.. the files on it were useless.
The only utilities that can encode wma files without the encryption are command line based utilities, right? I know average joe sixpack or Mr Aol Lamer Jr. won't know what to do with those. When was the last time you saw either of those kind of people use a command line and still have a working OS afterword? They're going to use the gui based utilities. Look at all the non-technical people use the gui-based mp3 all-in-one ripping utilities.
No, I do not see wma as an option until someone releases a 3rd party gui for the command line utility.
I'm a huge CDex fan myself, and it has always performed beautifully on .wav's and .mp3's, but it crashes on me (page fault) every time I try to encode an .ogg. I've tried contacting the writer of CDex, but the contact email address listed on the site is no longer a valid address, so I can't contact him. I've also tried oggdrop, which simply refuses to work as far as I can tell (nothing happens).
So I guess I'm going to grab a copy of EAC and hope it can do better. I'm stuck with hardware that can't run Linux, so I'm trying to find anyone who's written a competent and functional ogg encoder for Win98. Apparently, CDex and Oggdrop are not.
-Kasreyn
Kasreyn: Cheerfully playing the part of Devil's Advocate to hairtrigger
Now personally I use ogg/vorbis, but by this time there are more than enough posts supporting it. I'd just point out that maybe you should think ahead in terms of where the file format will be.
Why use an open format? Because in the end that's the only choice that makes sence. What program will you use down the road to play these things? With WMA MS owns the format, and thus can dictate who can play their files. What if they charge you a subscription fee just to use the program in the future? Who knows what they'll do, and they can do whatever they want - they have the rights to the format. You might also think about portability, and choice. If you don't like Winamp 7, you can use Sonique 5 or whatever. Chances are any player worth anything will have a plugin for ogg. With WMA, again it's up to Microsoft. What OS will you be using? It might not be MS or Linux. It may be something else entirely. Will you have to dump your collection because there isn't a player for that OS? I could go on and on, but you get the picture...
Link (or at least some evidence)?
What's really going on is this: using aggressive, fast-release peak limiting, musicians can get mastering engineers to push the volume of their CDs past zero. Actually, one popular technique is in fact clipping and then taking the overall volume down 0.2 db or so (to get rid of digital full scale values that can cause problems glass mastering, and with D/A converters)
Mastering engineers have been trapped in a jam comparable to clueful sysadmins being ordered to standardize on W2K/IIS: what's driving it is A&R reps and radio. Briefly, there are a lot of fools out there who figure their CD will sell better and get on the radio better if only it is louder than the next guy's. Sometimes that's even true as some of the radio program directors are also idiots who love horrible distortion and blasting loudness...
The trick is, there is NO one volume level that is 'the loudest' you can get out of digital. It's simply a tradeoff- how much distortion and grunge can you tolerate? It can be like putting a CD into a distortion box almost: look at modern music in a sound editor and you'll see a black ribbon because every sound is slammed to digital full scale. Look closer and it looks like the peaks get planed off with a surface planer. Sometimes this sounds like flat-out distortion, sometimes it doesn't, but it all more or less damages the richness of the sound.
At least with modern CDs, I'm not aware of ANY studios that put out CDs with peaks only going to part of digital full scale. The problem is in the other extreme- they pretty much all cover digital full scale peak to peak, but push beyond that in wildly varying amounts, which affects the RMS level. Some of the greatest albums in history were recorded with crest factor (amount peak is higher than RMS level) of 20 db and up, as much as 24 db sometimes (the Boston debut album). Some of your modern albums have a mere 6 db crest factor, or even less. If you put them on after the older album, they blast out your speakers and you have to turn it down (as the original poster said). Once you've turned it down, it's the same volume only sounds much lamer and weaker.
Which is all just a lot of information, no doubt, except that it is also the reason why your advice will totally NOT WORK in the slightest. Now, if you were talking about a 'normalize' function that looked at RMS volume it might be different...
A few years back there wasn an AAC encoder out there (Astrid/Quartex) which outputted rather descent quality audio. It was also a unique codec because it allowed for 5 channels of audio (the ISO standard, Astrids only supported stereo) and had comperable playback at 96kbps.
We coded a small GUI frontend for it and released it for the web to use. One month later we recieve a 28 page cease and desist from Dolby.
According to them, the Astrid/Quartex encoder was illegal and violating their patent on the AAC codec. The document stated a liscencing fee of over $10,000 a year for use of the codec.
So, as far as I'm concerned, AAC will be forever buried under the fat cats over at Dolby.
Great, not only is popular music smashed with limiting to within an inch of its life, but now Microsoft makes it policy to add another 3 db??? of smash just to beat other codecs in comparisons by untutored listeners?
There's actually a lot that can be done with doctoring the recorded values of FFT transforms. It's similar to spectral dynamics processing (in fact it IS exactly that). You could do it in playback with mp3, or ogg, or anything. You could build it into players as another sort of 'knob' to turn for those bored by EQs. But it is repugnant to have Microsoft building additional dynamics processing into their goddamned CODEC. My god, isn't popular music volume-smashed enough?
If you compare a good mp3 encoding ... to a WMA8 encoding of the same bit-rate and with the volume levels matched, mp3 will win out
I didn't realize WMA8 was compressing levels, but once levels have been compressed, it won't be possible to "match volume levels" and compare with original source or an MP3 as you suggest. (ie, either loud passages won't match or soft passages won't match)
With the quality of the latest RC3 release, Vorbis now sits on the throne in the low to middle bitrates, easily beating out MP3Pro and WMA even in the very low bitrates of 64kbps. The best part about it is that Monty has mentioned that he's still not happy with the quality at 64kbps and will still be improving it further. At middle bitrates of 128kbps, it is at least as good as the best AAC implementation. At the high bitrates, it still hasn't matched MPC, but it is catching up really fast. Whether Vorbis (a transform coder) can ever overtake MPC (a subband coder) quality in the future in the high bitrate arena (usually ruled by subband coders where pre-echo artifacts are nearly non-existant) is very much unknown, and probably depends on Vorbis implementing a really good anti-pre-echo system better than all the current techniques being used.
So therefore, for the best quality now, use Ogg Vorbis at bitrates of 160kbps and below. Above 160kbps, use MPC.
No, I am not a microserf, but neither am I a kneejerk linux zealot.
Somewhere, something incredible is waiting to be known. -- Carl Sagan
I posted to a Ask Slashdot a while back, and got some good feedback. The result was the following essay
Hope it sheds some light on the subject.
Toddlers are the stormtroopers of the Lord of Entropy.
WMA is just an audio-only subset of ASF - the file format spec is up at http://www.microsoft.com/windows/windowsmedia/WM7/ format/asfspec11300e.asp. You still need a Windows Media Audio decoder, though.
Umm... where the heck did you get the idea Compact Disc audio is analog? It is digital, silly.
Anyways, VQF has sadly gone the route of the dinosaur.
Legal just because you own the music or legal because the music is free? Technically I don't think that you are allowed to play your own CDs in a public place, like a school. ;). Also IANAL, so I might be wrong.
Not that I'm gonna stop you, though
Opinions stated are mine and do not reflect those of the Illuminati
Absolutely, DVD is lossy. If I took a DVD, decoded the content (which I think is always encoded with MPEG-2), re-encoded it with MPEG-2, and burned it to a DVD I would most likely have a worse copy than the original DVD. The process of encoding MPEG-2 is lossy.
Now it is true that I do not have to decode and encode every time I want to copy a DVD. I can use a non-lossy method of copying the digital data directly. This still does not change the fact that DVDs are lossy because the MPEG-2 codec is a lossy codec.
As for the "older and more established" definition, I could only find the following definition at dictionary.com:
Sapere aude!
You can keep your fancy MS Office, IE 6 can crash elsewhere, Quickbooks and Quicken -- you can have em;, games are for kids...But man if I could have 1 "port" from the evil demons it would be wma8 encode/decode. I am a bit of a music freak...and can not help think I get the best "bang for the buck" out of WMA's encoded at 64. The perfect mix of size and quality -- granted through headphones I did not mind the sound of mp3's at 56 back when 32 meg players were the rage....(So I don't have a professional ear...But until then Lame is cooking up pretty good 64's for me I guess.
(+1 Funny) only if I laugh out loud.
I wasn't talking about using MP3s in games (I agree that it is a good idea), I was talking about using OGGs in games.
"Reality is just a convenient measure of complexity" -Alvy Ray Smith
Some of my own informal tests (Sound Blaster Live and Sony MDR-G82LP not the best setup, but what i had available), i tested, ogg, aac (psytel and liquid audio/fraunhoffer), mp3, aac, vqf, wma7, wma8, mp3+ and more that escapes my mind (all lossy codecs), this is what i found:
aac consistently came at the top, but original psytel codecs gave a weird background noise, later versions fixed that though, it also had one of the highest decoding complexities
next came ogg vorbis, suprisingly this codec really delivered, there were subtle flaws (weird minor echos or treble highs just not sounding right, maybe wavelets will fix that) and it also had a very low decoding complexity
to microsofts credit, wma8 was quite good, coming mostly 3rd but it still had that weird swishy sound at times and it just sounded a bit synthesised (think 80s), but also came with a high decoding complexity
and a summary of the rest, vqf isn't worth a grain of salt, constant muffled sound, wma7 you need not worrya bout now there is wma8, mp3+ seemed to be an odd sort of tradeoff, not always getting better than mp3
these tests where done under windows, all codecs where forced to encode as close to 128kbps as they could, when i say high decoding complexity, i'm comparing that to mp3s
my final words would be, i'm looking forward to mp4 if it will take the best of both aac and vqf (dunno whats there in vqf, but hey they're giving it praise that i can't find for it), but its been indevelopment for so long and there aren't any available encoders (that i know of), it also comes with a high decoding complexity, in the meantime aac is very promising
ogg vorbis is what i choose, it has a favourable decoding copmlexity to mp3 and it still hasn't gotten up to its peak optimization so there's a lot of promise, i can't wait for this codec to be finished, it is just so great in every way, with sound only slightly worse than aac but a decoding complexity so much lower, it takes the crown, and there's still improvement to be done, note that i did these tests before there was an RC build, currently i'm builing a little decoder for ogg vorbis for my program, it sure got my attention
After collecting 60 Gb worth of mp3s, I switched to almost strictly shn format
over 2 years ago. Here is my reasoning:
1. Stick with a lossless format if you can afford the bandwidth and storage
space. Plan for the future, when bandwidth and hd space will be much
more plentiful.
2. I can definitely hear the difference between lossless and any compressed
format at 128 kb/s (that annoying wavery sound), and even at 256 kb/s (barely)
on very delicate passages and high-end speakers.
3. Also, if you want to reprocess the music (dehiss, dehum, equalize, normalize,
respatialize, etc) you experience a much more noticeable degradation in the
sound if you start with a lossy format.
4. shn is the standard format for trading music.
It is a lot less work to store in shn then have to decode and reencode every
time you make a music trade.
For lots of good links on shn format, see my trading page at
http://www.vsl.ist.ucf.edu/groups/vtb/TradeList
(Now that I've come this far, what the hell, trade requests here
.
;-)
I'm curious -- has anyone been able to metamod some of the moderations on the thread in question? Given that there have probably been more mod points expended on that thread than probably any other story, ever, it seems a bit fishy that at least I haven't seen anything about it. It might make the editors see things a little better if they got metamodded into oblivion.
How can we continue to believe in a just universe and freedom to eat crackers if we have no ale?
How about you learn what K IO slaves are before making such comments, hmm?
You either have something to do a digital to analog conversion in the CD-ROM drive itself and run a wire between the CD drive and your sound card which passes those signals through or you use bypass the CD drive's DAC and use a DAC on the sound card. Either way you're doing a digital to analog conversion and would be hard pressed to be able to distinguish a difference as a DAC is a DAC and either DAC you run the data through is converting the same data to something you can hear. In fact using the CD drive's DAC and a audio patch cable is MORE prone to noise and static than using the sound card's DAC.
I'm a loner Dottie, a Rebel.
I'm as mimsy as the next borogove but your mome raths are completely outgrabe.
I would jump to Ogg, no more mp3, no more anything else, if there were some l33tz0r Ogg hardware players... as it stands, I'm waiting on the very edge of getting an mp3 player, now that flash cards are moving at acceptable prices...
Please, someone show me a good Ogg player!
mp3s are a format of convenience for me. I spend 15 minutes encoding and storing, and then roll on from there. But I'd love to replace it with Ogg...
What we call folk wisdom is often no more than a kind of expedient stupidity.-Edward Abbey
You might find something like HawkVoice useful - it's a Windows/Linux LGPL voice-over-network API. It supports GSM, LPC, CELP, LPC10, plus some others.
-- Help Digitise the Public Domain at DP.
I would highly recommend reading these sites:
http://www.r3mix.net/ - Explains how to ge the best Quality from Mp3.
http://www.xiph.org/paranoia/ - How to rip a CD properly under Linux
http://www.exactaudiocopy.de/ - How to rip a CD properly under Windoze
With a 60Gb Disk sitting on a spare machine on a 100Mbit switched network I just use FLACs because its open source and available for Linux unlike Monkey Audio Compression:
Marilyn Manson.flac = 75% of original
Moonlight Sonata.flac = 42% of original
I just tried encoding Kosheen - Catch.wav at 64Kbit using Lame and Ogg with all other settings optimal. The Mp3 did'nt even come close to the Ogg. The Ogg was obviously gonna be a bit tinkly at 61Kbit average but the Mp3 was all muffled and horrible.
It's lossy because you lose quality from the original input. With lossless compression you can do something like:
compress as zip -> uncompress -> compress as gzip -> uncompress -> compress as rar -> uncompress -> compress as ace -> uncompress
and end up with the original file. With an audio example, this means you could transcode between LPAC, Shorten, etc. without any loss in quality.
With lossy compression, this is not possible. If you do CD -> wav -> mp3 -> wav -> mpc -> wav -> ogg -> wav, you'll end up with a really crappy wav at the end.
This has practical implications in that it makes transcoding unattractive. If for example you wanted to rip your CD collection to Ogg for archiving, but had an mp3 portable player, your mp3s in the Cd -> Ogg -> mp3 process would be of lower quality than if you had directly encoded the mp3s from Cd.
10 PRINT CHR$(205.5+RND(1)); : GOTO 10
You are wrong about that. You need a license even to play a CD or even the radio in public (I am not kidding)!
If you own a restaurant, and you want to play CDs or the radio quietly in the background, you need a license from ASCAP and BMI. JWZ talks about this some and all the crap he had to go through to do a webcast from his club. Here's a snippet that relates to what you were saysing:
One of the more absurd things about this system is the triple-billing that occurs. Consider the scenario of a retail store that has the radio on. That store is expected to pay ASCAP/BMI for the privilege of playing music. But here's what you get when you do the math:
Well, yes, you're right. But as far as "PCM files on audio CD's" are concerned: if you open an audio CD in data mode (i.e. as if it were a CD-ROM), you'll find a .CDA file for every track. The audio data in these .CDA files can be extracted quite easily, if you let your audio editor import them as "raw" audio data.
News and bla for computer musicians: http://lomechanik.net/
But I have no clue where to start looking for ogg music files. With mp3 there's BearShare (gnutella), Morpheous, AudioGalaxy, etc... but what about ogg?
~ now you know
The only utilities that can encode wma files without the encryption are command line based utilities, right?
Wrong. you can use advanced WMA workshop under windows to make wma files. Very nice GUI support conversion to and from wma, wav, cda, mp3 (if you have lame installed), ogg... does resampling of existing files and even has batch conversion. Will optionally delete source files once its done to save space...everything you could want, except a linux version
I reject your reality
Ok I have been corrected... I was trying to de-confusalize my explaination of normalization.. yes RMS is the best way to go and the program normalize does this in fact.
What I was trying to explain was that the mastering done today is horrible all the way around. Right now the high end audio shops are demonstrating HDCD and the sony HDCD system. everyone that listens to it comments how it's amazingly clear, wonderful sounding.... etc.... well back in 1987 I bought a CD that cost me $45.00 and sounds that clear. it is a Gold digital master of Supertramp's crime of the century album. and today it still blows away every other CD I have heard on any player in clarity,dynamic range, and overall quality. it is very close to this new HDCD format (in fact I had the salesperson use my wonderful CD in his player, he swore it was a HDCD until I had him find the copyright date, and try it on a regular Cd player.)
regular CD's cane be AWESOME, but the recording studios choose not to spend that extra hour to make it awesome.... because 90% of the CD's sold will be played on car stereos, boom-boxes, portables, and stereos costing less than $500.00
so yes, you are correct sir, but I still contend that the final mastering is still being done sloppy... and as you say, at the insistance of the buyer.
Do not look at laser with remaining good eye.