Slashdot Mirror
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."
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.
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!
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
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.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.
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.
Another thing... with high bitrate mp3.. when comparing between an original and the compressed version in a blind test, someone will be able to tell you they are different, but not which one is the original... becasue both sound good.
Well duh, then what's the problem? If they both sound good then you'll enjoy either one, so listen to the one that uses less bandwidth.
*sigh*
Or, even more interesting: peel a Vorbis file all the way down to the minimum quality. Concatenate the bits together in order. Now you have a file that you can play back, in its entirety, when it's only 10% downloaded. All you have to do is wait for the minimum quality version to download; from then on, the entire file is playable. It's just that the longer you wait, the more peels get added, the higher the quality... holographic audio downloading.
Of course. It all depends on what *YOU* are after, and how you enjoy music.
Many "Audiophiles" enjoy listening to how accurately they percieve their setup to be reproducing the original sounds... that's why they don't like lossy compression. It's not because it doesn't sound good.. it's because they are chasing accuracy. This accuracy becomes as important to them as whether or not they like the tune in the first place.
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
Actually, they are chasing precision.
None of them have a FUCKING CLUE as to how accurate their system is, or isn't.
I totally agree with you. But for different reasons (including some hardware MP3 decoder bugs) I did choose CBR 256kbps to backup my CDs.
;-) and I just want a backup. 256kbps quality cannot be picked up from the original (at least by anyone I know and by listening) on $450 headphones. I think that'll do it.
I'm using MP3 to backup my CDs. I've broken a couple of CDs recently (partly because of my 1 year old son
To try and answer everybody, you actually made a point I failed to address. MP3 should be used for LISTENING. The whole point of this compression is to remove frequencies that the global level of music is masking. Therefore, if you take an MP3 and apply a filter out of it (any kind), you will loose A LOT OF QUALITY, because the point of any filter is to modify the original and so some frequencies that were masked could have become audible.
Besides that, the main problem with MP3 is not a masking of frequencies but artifacts (I said the MAIN problem, of course frequency loss counts). The psy model used in an mp3 encoder will allow strong artifacts that could (and will) show up if you apply basic filters.
Write boring code, not shiny code!