Dial-Up Audio Public Listening Test Opened

CaptainCheese writes "Hydrogenaudio.org's Roberto Amorim just announced the opening of their 32kps multi-format listening test, intended to test the current 'dial-up' quality codecs. From the Announcement: "The formats featured are Nero Digital Audio (HE-AAC+PS), Ogg Vorbis, WMA9 Std., MP3pro, Real Audio and QDesign Music Codec. Lame MP3 is being used as low anchor, and a lowpass at 7kHz is being used as high anchor." These codec tests are unusual in that they adhere to ITU-R BS.1116-1. The test is open until July 11th and all are invited to participate. There's more info in the original test discussion, which indicates the originator is interested in 'testing formats working on dial-up streaming bitrates' - the test page notes: 'The real arena where codecs are competing, and most development is going, is at low bitrates.'"

Comments

What's the point?

[TommydCat]

I personally don't get the goal of a test like this. Listening at that low quality doesn't have as much commercial, and quite frankly, personal appeal as it did back in the 90's.

I've seen the double-blind tests done at 128kbps and again fail to see the point.

What I really want to see is a rating of codecs that are able to achieve DBT-proven audible transparency and see them rated in terms of storage space (thus allowing the VBR schemes to finally compete).

Of course FLAC would come in last (considering WAV is the 'source'), but can my high quality VBR LAME MP3 pass for the original and take less space than MPC?

Re:Get the news out to portable music player

[glassjaw rocks]

I really think that the mp3 format is going to be around for quite a while, mainly just because of how widely-used it is. Even though I know the wma format is superior to mp3, I still encode all of my cds to mp3, more or less out of habit.

Still too bandwidth intensive to be useful?

[Wild Bill TX]

No matter how optimized it is, won't it will still use too much bandwidth for dial-up users who actually want to do something else with their connection? All of the streams I ever tried to listen to, including the 8kbps ones, gladly used all of my available bandwidth. I don't know about anybody else, but I'm not interested in only getting a fraction of my 2 KB/sec max for browsing, using chats, or other tasks.

More interested in 32kbps speech

[Dr. Spork]

I am in the process of starting a project which needs accurate speech encoding at 32kbps. For now we're going with LAME at --alt-preset -b 32 -a --resample 22 --lowpass 6 -Z based on informal tests we did (ideas also came from here), but I'd love to see something more formal.

Notice all the different non-standard switches I had to use, which together help noticably. That's the sort of stuff you need to do to LAME before it produces acceptable results at very low bitrates. It is optimized only for 44.1KHz, so we should keep that in mind when we see the results. Notice now that none of these switches are being used for this test, so I'm almost certain that LAME will come out looking much worse than it is.

I would love for there to be a LAME-based encoder that is optimized for speech, low bitrates and sample rates. If it is made, I am prepared to re-encode all the readings that are (and are about to be) posted on my site.