Then we would, either via an automatic transcoding bot or via an upload enhancement to mediawiki automatically generate 40kbit/sec (dialup) and 96kbit/sec (broadband) versions from the lossless. These versions are what we'd link from our other projects.

First some background, why do we use compressed audio: 1) Uncompressed audio is huge and uses a lot of disk space for us. 2) Uncompressed audio would be an amazing waste of our bandwidth. 3) Uncompressed audio would be a huge burden on our readers, and would make clips longer than a few seconds inaccessible to users on dialup. Brion may stab me for saying it, but (1) is fairly unimportant. The way we are using compressed audio today is not very good from the perspective of (2), and (3). I haven't gone and exhaustively checked our files, but most of the Ogg/Vorbis files I'm seeing people upload are in the 160kbit/sec range. I believe there are four reasons for this: 1) An incorrect impression created by old data about MP3 performance. Old MP3 wisdom is that you have to be 160kbit/sec to be good. This was true five years ago. All cutting edge perceptual coders have results which are very good at 128kbit/sec, Vorbis better than the others, with results which are nearly statistically the same as the original on a test using trained listners.(http://www.listening-tests.info/mf-128-1/results.htm) 2) My recording is a unique and precious snowflake. I want it to be the highest quality. 3) You can go down but you can't go up, and I have broadband so it's fast enough for me. 4) Lossless files are bad for editing, less loss is better. Of these, *none* of them should be a factor is deciding what we send to our readers. Yet here we are sending these 160kbit/sec oggs to joe-average-reader, sucking up his bandwidth and ours. In terms of what we need uploaded for our own purposes, (4) matters. It matters a lot, because being able to edit the content is an important part of what we enable. However, even the 160kbit/sec lossy files fail here: Take a file decoded from a 160kbit/ogg and pass it through a 1500Hz high pass filter ('sox input.wav output.wav highp 1500' for those with real computers (tm)). The result will have obvious yucky artifacts. This isn't a bug, it's the behavior of a perceptual codec by design. The high pass trick is a pathological worst case, but it is true that even at 160kbit/sec perceptual codecs do not survive all forms of manipulation well. What I'd like to see us do instead, is to ask uploaders to send us losslessly compressed Ogg/Flac files instead. Lossless compression because disk space isn't totally irrelevant and to avoid people downloading insanely huge wavs just because they don't want to use the java player or install a codec. We already permit uploading these lossless audio files they can be easily transcoded to Ogg/Vorbis while preserving all metadata. Then we would, either via an automatic transcoding bot or via an upload enhancement to mediawiki automatically generate 40kbit/sec (dialup) and 96kbit/sec (broadband) versions from the lossless. These versions are what we'd link from our other projects. Considering the quality of most audio/video sharing sites on the internet, these would still leave us head and shoulders better quality. (Most youtube videos seem to be using 8 or 16kbit/sec mp3 for their audio, it's miserable but you don't notice if you're watching the video) Thoughts on this? On using a bot vs some native support? The biggest limit I see on this is that we don't have any support for subfiles, or bundled files... which means that this gunks up commons with two extra files for our audio.

What I'd like to see us do instead, is to ask uploaders to send us losslessly compressed Ogg/Flac files instead. Lossless compression because disk space isn't totally irrelevant and to avoid people downloading insanely huge wavs just because they don't want to use the java player or install a codec. We already permit uploading these lossless audio files they can be easily transcoded to Ogg/Vorbis while preserving all metadata. [snip]