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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."
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 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.
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.
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.
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.
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?
* 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...
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.
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
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.
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.
.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
* 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.
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
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>
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).
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?
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.
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
MP3Pro is mostly a marketing ploy. It has a 10 kHz lowpass filter, and then tries to reconstruct the upper frequencies based on harmonic extrapolation of the lower frequencies. This may be somewhat useful for low bitrates (say, under 80 kbps, for use in portable players). But the irreversible loss of audio quality makes this an inappropriate codec for the kinds of uses that I (at least) prefer: namely, files sitting on my hard drive on my desktop computer.
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...
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.
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 as mimsy as the next borogove but your mome raths are completely outgrabe.