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Bitrate Peeling with Ogg Vorbis
Yort writes "Thought this might be interesting to some audiophile /.ers - there's been some discussion on the Ogg Vorbis lists, summarized in the most recent Ogg Traffic, about "bitrate peeling". In short, it's where you can simply "peel off" the high resolution data from the ends of an audio stream packet to come up with a smaller, lower quality stream. Brings up a number of geek-cool opportunities."
This might just push net radio into the mainstream...
With something like this, a high-quality stream could be sent to a number of reflectors that reduce the quality and retransmit it for lower-bandwidth clients...
Could be kewl.
don't both progressive jpeg and mp3pro use this? a low bitrate basic stream is there along with additional sets of data to enhance it.
I don't know of any that play Vorbis files yet, but it would be very handy if I could take an OGG I had encoded at a high bitrate (for playback on my nice home stereo) and make it smaller for use on a walkman-type player for the gym or whatever.
I think another cool application of the technology would be to maximize the bitrate, not lowering it. You could download .ogg files to a...well, I want to say MP3, but it's not...OGG player at awesomely high rates. But then the player would dynamically peel the layers off when space ran out (actually, with my idea there would be no space, but it would not appear as such) to add new songs.
Wait a minute, I thought audiophiles always wanted to improve sound, not deteriorate it! Maybe the coders, like me, like this stuff, but my audiophile side is not interested
This is just sig!
You could do the same thing with simple transcoding but this could be a lot more scalable.
It would be even cooler if they could disperse the signal in time (i.e. across several packets) and just drop every n'th packet in order to create a lower bitrate stream without making it sound choppy.
"I have opinions of my own, strong opinions, but I don't always agree with them." -- George H. W. Bush
Bitrate peeling is a briliant idea, and would be a major win for Vorbis if they ever actually provide an implementation of it. It's something that the format supposedly supports, but right now it's still just a hypothetical application.
Let me know when they've got something working THEN I'll be impressed
my sig's at the bottom of the page.
this brings to light the mp3/ogg hardware compatibility issue... ie: no portables/car stereos/dvds/home stereos supporting ogg, even though it's a superior format... i'm scared that it's going to suffer the betamax syndrome...
personally, i'm converting over from 256kbps mp3 to 128kbps vbr ogg and i'm saving TONS of space and not really sacrificing any quality... does anyone know of a petition or something similar to get mainstream hardware manufacturers to include ogg support in their hardware?
similar to bit reduction decimation (look about halfway down the page for a quick explanation of decimation as an audio effect).
They say this would require some changes in the encoder.
It would be great if somebody made a tool for converting my old oggs to the new "peel-able" format. (Assuming, of course, this is possible.) That way I didn't need to re-rip all my CDs again. I'd just have to run the converter over the night. (or, to be realistic, weekend.)
Smaller and lower quality does not belong in the same sentence with the word audiophile. :)
Cool idea though.
Q.
It would be cool if you could "wrap" a few extra bits to the end to add quality. I know that it will never work but a cool idea nonetheless.
----
Go canucks, habs, and sens!
Lately I've been finding all I can download off P2P programs like Direct Connect and Furthurnet. Its mostly live shows, and they are all in .shn format, which is a lossless compression format that restores to the original .wav file.
These communities shun both compressed files like .mp3 and trading anything that has been released commercially. What you do get is great recordings of live music from bands like U2, DMB, Grateful Dead, etc., all ethically traded and in their full audio glory.
The audiophiles I know pretty much don't listen to mp3, ever.
No, Thursday's out. How about never - is never good for you?
Bitrate peeling is not a good idea because it seperates the audio signal, with hi frequency data being seperate from low and mid frequency data.
I'm not sure if it's frequencies that are being separated here - the idea seems similar to wavelet image compression, where refinements to the original data arrive over time.
Note I know nothing about Ogg, so it probably isn't based on a wavelet approach, but the idea sounds more like you get low resolution data (covering the full frequency range) followed by higher resolution data. I.e., you're increasing the sampling rate over time - but the range of frequencies you sample is the same regardless.
Nae bother
I always kind of felt the first and third bit of every byte were kind of unnecessary. I'm not to fond of the F6 key either!
I don't think that's how it works.. from what I understand, it simply removes bits... ie:
I = one bit
----A 128k stream----
IIIIIIIIIIIIIIIIIIIIIII
----A 96k stream-----
IIII IIII IIII IIII IIII
It simply removes some of the bits, not any specific freqency of the high/low/mid.
and it looks to be an impressive streaming method
Squid
But what do I know. I'm just looking for anonymous gay sex.
"Beni Cherniavsky mentioned a very intriguing counterpart to bitrate peeling. If you have a peeler that saves the bits it chopped off, you could reconstitute the higher quality files by adding the missing bits to the lower quality file. This idea could lead to a music download service where you can download a low quality preview version of a song, and if you are interested, download the missing bits to make it a high quality version."
Or, Slightly modified, You could share all your high quality oggs on a P2P network, and have your client peel it down to 'future-legal-to-share' low quality files.
Pain lasts, kid. Its how you know you're alive. Sometimes I think this growing up thing is just pain management-TheMaxx
Were Bitrate Peeling and Busting Pimples bands?
The same one that came up with Leto2. :P Jeez man, it's just a name.
slashdot!=valid HTML
Yes they are in the same genre as Smashing Pumpkins.
Quit Slashdot Today!
Sounds more like it would be unique to OV, if they implemented it.
The point is, nobody does it now. Perhaps this is because there's really no need for it. Consider the list of "very sexy applications:"
- In a streaming situation, the server would store only one high quality stream, and dynamically peel it down to the client's bandwidth. Not useful. If you instead stored a hundred separate files, each optimized for its bitrate, with each being half the bitrate of the previous one, you'd still have a set of files less than twice the size of the largest file. Plus, you'd have no bit-peeling overhead. If you're streaming 100GB audio files, maybe there's a benefit, but if you're doing that, you can probably afford a second 100GB file for all the smaller files.
- You could store high quality Ogg Vorbis files on your PC for your Audiophile home theater setup, and peel them down to "good enough for lousy headphones on a noisy train" portable files. You can do that now, without this high-tech. Such low-quality files could be easily made from the original OV high-quality files, without much extra artifacting due to the re-encoding. And again, how low a quality are you willing to accept, if you're going this far anyway? Wouldn't you just buy a higher-cap memory card?
- Download a low quality preview version of a song, and if you are interested, download the missing bits to make it a high quality version. Another non-benefit. Suppose the full file is 10 meg, and you download a 1 meg sample. Are you really going to opt to download the 9 meg "patch" file, rather than the 10 meg complete version?
A clever idea, and it sounds cool on the outset, but it seems to me that this is a solution seeking a problem.That's exactly what the linked summary of the mailing list says! :)
PATENT IT!
Before some lowlife corporation does.
This concept probably came from wavelet compressed video. The different layers within the wavelet stream contain different frequencies ranges and can be easily separated out for various purposes. You can for instance grab just one frequency band to use as thumbnails or proxy images instead of recontsructing the entire image and scaling. The problem with applying this to audio is that the human ear is a lot more sensitive to data loss in audio than it is in video. If you simply peel off high resolution data you would lose the most important, constructive portions of the audio signal. If this was a viable solution we would have long ago settled for hardware bandwidth filters rather than sample rate conversion. What history is told is that it is better to give up data temporaly than it is to give up frequency response.
This is exactly the killer app that Ogg needs for acceptance: a program that syncs songs to your portable player at a lower (user adjustable) bitrate. Even better: You pick out X number of songs. Each time you add a song, it re-calculates what bitrate to shave them all to, to maximize the bitrate used, thereby using all the RAM on the player but getting all your songs in.
I can't wait til this one hits.
That should be, "All your base are belong to Bortman." Please fix, OK? Thanks.
And well done on the fine first post.
Keeping All Your Base parodies correct since AD 2002.
If you read many of the audiophile magazines such as Stereophile, etc., you'll see reviews of equipment such as external DACs for CD players as well as many high-end CD players. Among those there is the legendary Linn Sondek, a CD player which retails for around 21k.
Why would you need a 21k CD player, you ask? If a CD player is playing back an exact digital file, than shouldn't all CD player's sound the same? The answer is simple: let your ears be the judge.
I was initially skeptical when I was first shopping for stereo equipment, but there is a world of difference between a consumer CD player obtained at the chains like Best Buy, Fry's, etc. and an audiophile CD player. The difference is primarily in the level of clarity or resolution that you can hear from a quality CD player. The difference is subtle yet dramatic, you can hear instruments and detail that you simply could not make out before.
To make a long story short, the quality of mp3s is typically even below that of a CD played on a cheap consumer CD player. No "audiophile" will listen to them as a primary audio source. That said, I have an mp3 player that I use when running and I have mp3's on my computer. Everything has its place, but the place of mp3's or ogg is not in audiophile stereo systems but in the world of music sharing where file size is a critical issue.
Musi withou hig frequenc is lik wods mising leters.
-braxton
Just jam a crayon up you nose. They you will understand the joy of paying $100.00-$500.00 per foot for speaker cable...
STOP using kewl. You sound like a friggen 13 year old script kiddy.
of a Slashdotter comment that twisted an old expression...
"Those who sacrafice sound quality for hard disk space deserve neither."
Cool idea to throw around though.
Why, o why must the sky fall when I've learned to fly?
One thing that hasn't been discussed here is that a lot of people feel that Vorbis is transparent at something like quality setting 4. Other people think it's transparent at quality setting 3. Others think it's great at 1. I release my stuff at 4, but bitrate peeling will let people peel those down to what sounds good to them. Maybe they want to monkey with it, and maybe they don't, but the option to do this without re-encoding is sexy.
It's not just a 'chop it down for modem folks' thing, it's also a letting people choose for themselves situation that I think is more important.
Features are cool, but features that give people options apart from 'use it or not' are even cooler.
That's it for me. Please donate to Xiph.org, and then go listen to some tunes. Enjoy!
Sony and audiophile are usually not mentioned in the same sentence together.
SonyES what you jammin' on Stanley?
It's overpriced for what you get friend.
In theory your first point makes sense if your tight on space. On that point though, where does it become economical to have the trade off of spare (presumed) server cpu cycles for bitpeeling and hdd space for pre-peeled files? With the current state of X86 hardware, or even PPC hardware, one would think that it would be easier to use space CPU cycles, but I havnt done much research into this. An interesting thing that someone could do and throw up as a little chart on the Vorbis FAQ site for those of us interested and too lazy to do it themselves. ^_^
We don't need an "overrated" so much as we need a "you completely missed the parent's point, dumbass..."
I've been asking for this one for quite some time
now!! I was waiting for this feature before I
re-rip all of my CD's.
Now, if they can have it go all the way up to
lossless compression and include this peeling,
that will rock.
I agree... you're lowering the resolution, not the sampling rate. Sounds like instead of taking 16bit samples, you drop the LSB and send 15bit samples at the same rate.
Intel's old Indeo v5 codec (back when they still supported it) did something similar. When you downloaded an Indeo stream, they sent you progressively higher quality versions of the same stream. People with slow connections got a low quality movie that started up almost instantly. People with higher speed connections got a movie that started out blocky and jerky for about a second but quickly sharpened up.
...to avoid crying.
I dunno. Works for onions.
What about LIVE streaming? Should the audio source encode 10 different versions at once, and send double the bandwidth to the repeaters? Bitrate peeling is a great benefit for live streaming, it will reduce the upload to a single stream and take processing power requirements away from the encoder.
This idea of peeling away layers is great for if you have a high bitrate file and want to use it on something of lesser "quality". What I don't understand is why these high quality, lossy codecs aren't good enough, ESPECIALLY for live shows. Maybe I'm missing something but if someone is taping a show off a microphone onto some digital device of some type, aren't you just losin quality there anyway? Even if you're patched into board you still have loss AND its a live show so like you could tell the difference anyway.
- gtaluvit (prnc. GOT-tuh-LUV-it)
I doubt that. They sound much better than MP3's at lower file sizes. Although I highly doubt that MP3's will dissapear entirely for some time, I think you'll start seeing vorbis living among them in large numbers in the near future. Many software programs are starting to add vorbis support with no signs of slowing down, and Apple will soon have an iPod with Vorbis support. It's just the beginning of large things to come...
Scalable techniques like this are very cool, but hardly novel.
MPEG-4's scalable profiles provide a similar effect (albeit in the other direction, with enhancement layers). Some of the higher end audio codecs (beyond CELP and AAC), like ER BSAC (Error Resistant Bit Sliced Arithmetic Coding) do exactly this. The idea in this case is that the server will in real-time only provide as many bits as the connection can currently provide. Very nice for wireless.
QDesign's QDX format does almost exactly what is described for Ogg, with arbitrary bitrate peeling down to the 1 Kbps level. The idea is that you could copy as much data as you want to your mobile player, and it'd dynamically thin to the data rate that would fill up your device.
And still image codecs like JPEG have used progressive modes for years, where additional data adds more detail to the image.
My video compression blog
The difference is that SHN has gotten 18+ months of exposure as compared to the 2-3 that FLAC has gotten.
Granted, I understand where your coming from, but thats why you see it everyone, as such, in my mind, it gets the job done and a lot of people also believe that. Same reason (well, one of) why Ogg isnt well saturated.
We don't need an "overrated" so much as we need a "you completely missed the parent's point, dumbass..."
Okay, if each one is double the one before, that means you'll have a 2^100 ratio between lowest and highest data rate. Thus, if your lowest is 1 Kbps, the highest would be... Not going to happen that way.
Also, you assume the sweet spots are 2x the one before. In fact, jumps of more like 1.25 are likely to be optimal (albeit with a lot fewer jumps!).
My video compression blog
A-men. Ditto for Bose.
slashdot!=valid HTML
It would be great for multicast-type situations (including, but not limited to IP multicast.) You could send one high-quality signal out from a central point and then shave off bits to fit the stream down to the quality needed by the end-user.
For instance, users on a 56k modem could listen to the same multicast stream as a broadband user (ie, no need to send out multiple, separate versions from the source)-- this assumes the presence of routers (or conversion boxes) capable of doing the peeling as needed.
All in all, very useful.
And as others have noted, audiophiles do not care for ANY compress format, OGG or not.
You know you have a nice system when everything you play on it sounds like crap :)
Note that the difference between both of the above scenarios and the one you proposed is that here we're talking about network bandwidth rather than storage space. While it might be cheap to keep many different versions of a file on a server's hard drive, bandwidth is frequently a more precious commodity-- especially if you're talking about live events where constraints are generally even greater than under other circumstances.
Could it possibly be used to progressively download an Ogg file and begin listening to it before it's 100% done? For example, have the 32kbit quality at the start of the file, then next have the next 32kbit (starting at the end of the file), then the next 64kbit, etc...
Luke-Jr
Your mom jokes about you.
I suppose it's possible to emulate most of this in IPv4 multicast using several streams, but it's not as automatic.
He was bragging about his expensive (and therefore wonderful) setup, when he mentioned having "super-high quality" digital cables, which cost him $2,000. I asked him how much "low quality" digital cables cost, the answer? About $15!
I asked him what the difference between the two was, he claimed that the high-quality digital cables gave the music more "body"!
It may have been cruel of me, but I just couldn't help but explain what digital actually meant.
The moral of the story? That much of the audiophile community are simply the blind leading the blind, pseudo-techie alchemists, who assume that expensive means better.
"On my Complainophile Sony Type-R Limited (bling-bling) audio set up, 32kbps sounds like ass because I didn't realise I could encode at higher bitrates. Except Windows Media because there were no options... then it scanned my hard drive, told me I had really poor taste in music and automatically deleted all my .mp3 files.
Bitrate peeling is not a good idea because I say so and need to look important. Transcoding is such a better idea because it introduces no artifacts and takes a much shorter time to do. Not that you should do that because nobody uses a narrowband internet connection anymore, at least nobody that I care about. Comparitively RealAudio is great! No, really! I enjoy giving my personal information to someone just so I can listen to commercials and kill pop-up advertising before it asks me if I want to upgrade, then crashes my computer!
While I conceed it's possible.. nobody else knows what they're talking about! Um... look over there! (exits stage right...)"
Yes. Thanks. Really persuasive argument.
duh. double blind audiophile tests of 256kbit/sec mp3 vs the original CD source have proven that no difference can be heard. an audiophile is someone with too much money for toys who has to show them off and treat others like dirt to make their testicles feel larger.
ever see an audiophile woman? nope.
there you go.
Vorbis peels YOU!
Regards,
Marvin
AVI files have trouble keeping sync with VBR audio and Ogg/Vorbis only does VBR. I'm actually working on modifying the Vorbis codec to emulate CBR with average bitrates and presently I have to drop full packets when the avg bitrate goes too high. This way I could just drop a few bits from each packet to pin the avg bitrate to a specific number.
The peeling is designed so that what's lost is frequency *resolution* rather than frequency bands, of course encoders may decide that the perfect peeling profile will lose high frequencies, but that'll be to preserve frequencies closer to the middle of the hearing range.
Anyway.... the words Sony and Audiophile don't go together.
There aren't ANY free lossy codecs that can do bit-peeling right now. Some non-free codecs allow you to overlay data (like progressive JPEG), which is NOT the same thing. That's just like transmitting deltas at higher compression rates, which could be done with a simple side-band for any existing method, MP3 even.
The only option is transcoding, which compounds compression errors (decode, reencode). I often wished the MPEG group would have been more intelligent in the design of their bit-allocater so that you could "thin out" the quantization of the power bands by looking at the "right parts" of the MP3 frame. Alas, this is not possible.
But the Vorbis designers have made this possible, thus making it possible to have high-quality and low-quality versions derived from the same source file without additional processing. I imagine you have certain restricted choices, due to how quantization information is bundled up/packeted. But it isn't just sexy, it would be stupid to NOT DO IT. It takes just a little forethought on how to lay out the information in a hierarchial fashion. What makes anyone think that this is any harder then decoding/reencoding. I guarantee it has a time complexity on the order of a straight copy.
Hell, formats like SHN and FLAC can do it, just substitute short codes for long codes at a certain rate; it'll add a bit of wide-band energy on decode, raising the noise floor in proportion to the space savings you gain.
So anyway, "poop on you" to all these wanna-be audiophiles/slashdot-know-it-alls who don't no a good thing when they see it.
Don't like it, keep sucking on that Layer III.
THIS THING CAN TURN ON A DIME, MACROSSZERO STYLE ALSO FUCK BETA, ~NYORON
Speaking of such, are there any Ogg-supporting portable players, or players in development?
(Granted the Hardware Support page at Xiph has some info, but I'm curious if there's anything else known)
Alex Bischoff
HTML/CSS coder for hire
FlashPix is similar with OLE Structured Storage thrown in to get Microsoft to participate (much to the agony of anyone who's ever tried to write a FlashPix file parser). PhotoCD is similar except I think it might not be tiled - Kodak was a major partner to Live Picture and even though the original LP format would have worked fine, Kodak wanted something proprietary :-/
It stores the original resolution, only in tiles whose size are about what would fit on a typical monitor. Then it stores half that resolution, tiled again, and so on. I think there are six levels of decimation. The total file size is about twice the normal full-resolution file.
The advantage of this is that you can pan and zoom to any portion of the image quickly. Only a modest amount of scaling would be needed to get to the view the user selected.
The really sexy thing about Live Picture (a high-end grahics editor) is that it never applied time-consuming graphics operations to the full image. Instead it would only render what was necessary to show the results to the user on the screen.
All of the edit commands were saved in a display list, and re-rendered every time you changed the view or edited in some way. You could save your display list in a file that linked to the graphics, and in effect have infinite undo that could be continued across launches of the program.
Each kind of operation you could do to an LP image was a layer - there were monochrome paint layers, multicolor paint layes, distortion layers and so on. You could composite images with image insertion layers. I understand Adobe got the idea for putting layers into Photoshop from Live Picture.
The final rendering to a TIFF file was time consuming, yes, but could be left until the end of the day and ran as a batch job overnight, or offloaded to a separate machine.
This made Live Picture a very complex program to work on. It had about 70 MB of really arcane C++ source code at the time I worked there in 1997.
But it made Photoshop look like a kids toy, because it could easily and very responsively handle the compositing of a half-dozen 200 MB images on a 150 Mhz PowerPC 604 Mac 8500 with 32 MB of RAM - I had machines like that both for my main development machine at the office, and coincidentally I had an identical machine at home (which I'm typing on now, although it's been upgraded several times).
While Live Picture as a company had great technology, unfortunately it failed to compete as a business against Adobe. Read more about it in:
-
The Valley is a Harsh Mistress
After its bankrupcy, Live Picture was acquired by MGI Software of Canada. Later MGI was acquired by Roxio, the Adaptec spin-off that publishes toast and easy cd creator. Roxio publishes a bundle of inexpensive graphics utilities, but I don't think Live Picture is included. That is a sad end for a powerful graphics application that once retailed for $4000.Request your free CD of my piano music.
Looking at your example of storing the variable bit rates as seperate files as an example, let's do some theoretical math:
Original High Quality file: 10mb
3/4 Quality file: 7.5mb
Half quality file: 5mb
Quarter quality file: 2.5mb
Total for all variations without peeling: 25mb
Or, store the High Quality 10mb original only and dynamically peel. Savings, 15mb (1.5 times more files)
No need te reencode a new file for each device (and some of us have many!)
Just FYI
You can have it fast, accurate, or pretty. Pick any 2.
Doesn't killer app imply that there is something new about the idea? Windows Media Player 9 (beta) already does this for copying to portable devices with WMA files.
but what you just said has nothing to do with whether a 128kbps Ogg sounds like a 256kbps mp3.
It's commonly observed that oggs of lower bitrate compare to mp3 at higher bitrates.
I'm not going to get a player with bitrate peeling until it PLAYS OGG VORBIS!!!
There are a number of problems with the parent post.
1. Keeping hundreds, or even ten separate files as described, each half the size of the previous, is not plausible. I'd assume the author was a troll, but since no one else has mentioned it, perhaps the obvious fallacy with that idea is slipping past even the sharper readers. A 10MB file can be split in half at most 23 times before it is only 1 byte long, far fewer before the quality level is unacceptable. Secondly, the idea described in the article, provides for dynamic bitrates, not simply half the original bitrate. To provide even similar functionality, one would need files in ranges from 1MB to 10MB in relatively small increments, totaling well in excess the "twice the size of the largest file" as suggested. Even so, this would be deficient in that the bandwidth could not be throttled mid-stream.
2. Second, decoding and re-encoding the same file with a different bit rate will almost certainly result in poorer quality than the technique described. The safer, more straightforward solution, is to perform reduction operations on the transformed data, rather than the decompressed waveform. Otherwise, amplified artifacts from the original compression will be present in the new file.
3. Third, the strength of the poster's argument lies entirely in the choice of ratios. Downloading a 5MB file rather than a 10MB file leaves only 5MB remaining. To paraphrase, are you really going to opt to download the 10MB complete version when your software can download the remaining 5MB in half the time?
There are a number of problems which bitrate peeling address, not the least of which are 1) reduction of storage space as described previously, 2) dynamic bandwidth regulation of audio streams for streaming radio, future cellular phones, VOIP, and network appliances running on congested networks, 3) file size reduction without transcoding, 4) user-specified bandwidth on demand, 5) automatic preview generation from source without any extra administrative overhead.
I'll even add my own... the ability to download a very high quality file and start listening to it immediately at lower quality without interruption. By the time the file has played through, the download may be only 50% complete. If I decide not to continue with the download, I have wasted no more time than that necessary to listen to the file. If I want the file, I have only 50% remaining.
In some ways, this is similar to the rationale behind interleaved images, except that it is unlikely that you will need to listen to the same file repeatedly at progressively higher bitrates. Nothing prevents this of course.
-Hope
Proof again that p0rn drives the Internet. What's next lap dancing?
First Post gets YOU!
What was your testing method? Was it double blind? Did it include volume randomizing? When we do double-blind testing in my house, many strongly held opinoins (you can easily tell X from Y blah blah blah) get shot down in a hurry.
I'm no audiophile. I think the long protracted discussions about cables, wiring and shit are a joke. (And convincing me to spend $40 on a OPTICAL cable to run 6ft ain't going to happen, I'll take the $15 cable. ($15C)
But even *I* think 128k MP3s really aren't that great when compared to the original music.
No comment on ogg at 128, never use it.
...I love it.. they'll bitch about mp3's being of lower quality when cds are already bad enough as it is. digital formats can never record something properly.. you know things are gunna be truncated. Some of us still use the superior black flying saucers your parents might have listened to.. unfortunately its true that they can get scratched too easily and all the rest of it but if u have a decent turntable and all the rest of it you should be able to play 'em over and over and over again and it'll still be better than any cd.
which is why I ASKED if the person honestly thought that OGG was twice as good ?? I really don't care on my end which format I use, I just want the closest to true sound. If an OGG track was actually as good as an MP3 of twice the bit rate it would be time to begin re-ripping things to Ogg Vorbis. As for space vs quality that is ALWAYS going to be an issue...
errr....umm...*whooosh* *whoosh* Is this thing on ?
The RIAA can replace the obsolete bits of each packet frame with
their latest DRM trojan code that makes very sure that you don't
accidently have copy-righted music on your computer.
too little substance. While it might be nice, as well as a good idea, I've been waiting EVER SINCE OGG RELEASE for SIMD acceleration of encoding. WTF IS WRONG WITH YOU PEOPLE?!? I should use it, because of this or that or blah blah blah. Yeah go ahead and mark me for flamebait or trolling, but the cold hard fact of the matter is this: noone will take you seriously (read mainstream) when your encoder takes roughly TEN TIMES LONGER than (for example) gogo. Other 'commercial' encoders are even faster...
That's how DTS was able to add a discrete surround channel (DTS ES) without causing problems with older receivers. Dolby can't change their header without breaking backward compatibility, which is why their extra surround channel (DD EX) is matrix encoded.
* As is generally the case, my opinions do not reflect those of my employer.
I think that the iPod does a very bad attempt at this. MP3s, encoded at 320k play back really badly on the iPod, far worse than 128/192k ones. I suspect that the iPod hardware hasn't enough horsepower - and it is discarding bits that it cannot decode fast enough. The MP3s sound fine on the pc, or decoded to wav and then played back on the iPod. But played back (as MP3) on the iPod, the result is dismal - there's a 5Hz "wobbling", rather like a steel band, and lots of distortion. (Apple won't help, but I have replicated this problem on multiple setups both Linux and OSX - it would be interesting to see if any /.ers have seen the same thing. You need a good recording of a classical CD with very large dynamic range eg Mahler 8, part II to demonstrate it - listen to the quiet bits.)
[I have some demo files, but can't link them - I'll get slashdotted off the net !]
...but how many vacuum tubes do you have in your system?
The true measure of an audiophile.
"And like that
-1 Troll???
That's a +1 funny at least.
Now I got it..thanks
errr....umm...*whooosh* *whoosh* Is this thing on ?
Okay, someone start the tubes vs. solid-state battle already!
*moan*
Get the nifty new ogg player and stick a 4mm jack into it and you have an ogg player thats also a tiny linux machine. Love mine! Also plays mp3's and mpeg1, etc. Mine has about 300Mb right now.
How does my quip "iPod'ers with 20 gig drives wouldn't lose sleep over it" become quoted as "20G should be enough for anyone!" in your post?
You must work for CNN.
You can buy the OGG player for like $15 and use one of those cassette->cd adapters. You know those things that look like a cassette but have a wire running out. Just plug it into a Zaurus instead of a CD player and you have a mega-audio OGG player gizmo. Mine has a 256Mb SanDisk SecureDigital card so it has tons of storage space. These gizmos rule. They really are pretty buggy and weird, but nothing a slashdotter shouldn't be able to handle :-)
Peel layers from a data file thin enough that no layer is independantly recognizable. Scatter the layers throughout a P2P system. Now no individual user possesses the whole file, but all users can reconstruct a working copy when they want to play it.
That Jesus Christ guy is getting some terrible lag... it took him 3 days to respawn! -NJ CoolBreeze
Since you clearly used "preview" before posting your message, you could have just as well used html.
Compare these:
(no space between "ht" and "ml", of course: thank/. for that)
<a href=" ">click here</a>
What the fuck.
Because this Jackass quotes a concept directly from the article, he's fucking "Insightful"
I read the article earlier today, then came back to view comments, and noticed that this was the latest post at the time, and was ~ the 10th post, and thought to myself, "This will probably be moderated up, how pathetic."
If I wasn't capped long ago, and cared enough about karma, I suppose this would be a great strategy. Read the article, and then posting clearly enough that I'd read the article, take the coolest concept in the article and pretend I had independently thought of it on my own.
Fucking pathetic.
Packet filtering has been done for video many times.
Its an interesting variant where network filters
read frequency band tags in the headers and decide
whether to keep it or not (see for ex.
wavevideo
It's not only for storage but mainly for online
applications, it works especially well for
multicast traffic.
"Thunder is good. Thunder is impressive. But it is the lightning that does the work."
When I explain to people different audio codecs and file formats, I usually use an analogy to graphics:
Now I'll be able to also use progressive JPEG analogy, very cool! But seriously, I just love the idea that I'll be able to have one copy of everything at -q5 or -q6 (hell, even -q9!) and copy as many bits as I need from that. Extremely cool in my opinion.
I think people at Xiph.org are doing a great work. In fact, I'm starting to re-rip my CD collection to save them as lossless FLAC files (in Ogg format, of course!) to be able to encode them as Ogg Vorbis 2.0+ in the future. Almost all of my music (i.e. everything I have on my own CDs) is encoded with older oggenc from something like a 6-12 months ago, and now with the bitrate peeling I know that sooner or later I'm going to encode it all once again, so I'm starting to rip it now.
By the way, some comments say about how this bitrate peeling in Ogg Vorbis would be cool for speech compression, VoIP, speech streaming, etc. Don't forget to check out Speex project, which is now part of the Xiph.Org Foundation. From Speex website: "The Speex project aims to build a patent-free, Open Source/Free Software voice codec. Unlike other codecs like MP3 and Ogg Vorbis, Speex is designed to compress voice at low bitrates in the 8-32 kbps/channel range. Possible applications include VoIP, internet audio streaming, archiving of speech data (e.g. voice mail), and audio books. In some sense, it is meant to be complementary to the Ogg Vorbis codec." Does anyone know if bitrate peeling will also be used in Speex?
root@aio:~# nmap -sX -iR -p1- # Ho, ho, ho! Merry Xmas, everyone!
really.
I was pretty convinced that "bit's were bits" and that jitter was just elitist hoodoo mumbo jumbo, but here's a couple links to jitter articles that made me question my views a bit:
Jitter on various transports [Stereophile mag]
Informative page by John Risch (DIY cable / EE and acoustics specialist[if memory serves]) link on cables and jitter along with links to more articles about jitter and wire design.
According to acquaintance of mine whose parents ran a high end audio equipment store, jitter does make some difference, but it's not that big a deal. As for me, I've only imagined that I found differences in some interconnects, but I wanted to do AB tests with higher end equipment under stricter test procedures before making any judgments I'd put my full credit behind.
Hey, that reads remarkably like something I have read before. Oh wait, that was in the article itself.
Wouldn't it be nice if people would actually read the articles they commented on? Ah, well, maybe a next lifetime.
.sigs - is there anything they can't do?
A-bloody-men!
All this audiophile nonsense, what with the insanely expensive power cables and such, smacks of simple elitism. "Look, I can spend twice as much as you! Hence, my audio is better, and I can appreciate it more than you can!"
I consider my low standards for audio a blessing. (Especially since I can't afford 'good' equipment.) If I don't hear a problem, then I define my solution as "good enough". It's cheaper, easier and I bet I get just as much out of my music as anyone else does.
And besides, if audiophiles were so nuts about quality, why don't they do more double-blind tests? Scared they'll discover that their A$240 power cable doesn't present an improvement over the freebie they picked out of the trash bin?
I swear, these people are crazy.
--grendel drago
Laws do not persuade just because they threaten. --Seneca
You are a sad, sad man.
The player could strip its own files on-the-fly. If you wanted to add one more song to the player, it could strip a tiny bit off of each song (or maybe just strip the higher bitrate songs), and make enough room for the new file.
HIV Crosses Species Barrier... into Muppets
As reported by Palm InfoCenter, "A group calling themselves Aerodrome Software have released a public beta of an Ogg/Vorbis media player for the Palm Tunsgten T handheld. The player inititally supports only ogg/vorbis encoded files, a new open source audio format, but promises to have mp3 support in the near future."
I'm curious as to whether they used the integer-based "Tremor" code to achieve this.
Alex Bischoff
HTML/CSS coder for hire
peel off the high resolution data from the ends of an audio stream packet to come up with a smaller, lower quality stream
Newsflash: removing data from a packet makes the packet smaller, and as a totally unrelated bonus, the content of lower quality! This obviously deserves space on the front page of the world's largest geek magazine.
Something similar to progressive JPEG encoding might be possible, get the full song in 30 seoncds? and then spend the next 5 minutes watching the quality slowly improving until you have the full thing?
However, because the source is available, there have been versions of it, modified by others of course, that go against the spirit of peer to peer file sharing. This, among other reasons, has led to some hub administrators banning the use of this client. If that's the case for your favorite hubs, you might consider finding new ones.
I'd say the difference in quality and stability is similar to eDonkey vs eMule.
Independent artists and labels are free to license their music however they want.
Even if the performer is not with a major label, the songwriter still gets a fixed amount per copy distributed. Performers who write their own music have no easy way to verify that they didn't accidentally infringe the copyright of an existing work, which George Harrison found out the hard way.
Will I retire or break 10K?
<a href=" ">click here</a>
"Sorry, links to this site from Slashdot are disabled."
Will I retire or break 10K?
it simply removes bits
It's a lot more sophisticated. It's not lowering the sample rate, and it's not droppping the least signifigant bits of the samples.
It's hard to explain without getting into the complex math of Fourier Transforms, but every bit has information from the entire file rather than merely a small piece of information from one spot. The bits are naturally in the order of importance, and dropping the bits at the end drops the least important information everywhere.
-
- - You can't take something off the Internet! That's like trying to take pee out of a swimming pool.
Hello,
The Sharp Zaurus can play Ogg Vorbis since it runs Linux.
I use my Zaurus as an Ogg/MP3 player everyday and it works quite well. Bitrate peeling will allow me to store more music.
Interestingly, both Ogg and Vorbis are key characters in various novels by Terry Pratchett.
I don't know if this is the real source of the name, but it seems more than a coincidence to me.
Unix gives you just enough rope to hang yourself -- and then a couple ... We make rope.
of more feet, just to be sure.
-- Eric Allman
-- Rob Gingell on Sun Microsystem's new virtual memory.
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