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Low-bandwidth Net Radio

Posted by michael on from the bit-conservation dept.

An anonymous reader writes "Slate has an article about Internet radio stations that use the aacPlus codec from XM satellite radio instead of MP3. Some of the ones they link to sound pretty good even at 24 kbps."

3 of 143 comments (clear)

  1. Ogg streaming seems pretty good by Xenna · · Score: 4, Interesting

    I'm not an ogg-head but I was pleasantly surprised by the quality of 32 Kbit ogg streams a while ago.

    http://www.virginradio.co.uk/thestation/listen/ogg .html

  2. i dont get it by hasst · · Score: 5, Interesting

    I really don't see the point in this article. I've read it, and then re-read it. They are comparing a "new" codec with MP3, Windows Media 8 and Real Media 8. The document in which they present the "clear winner" is dated June 2003. In my time that's more than a year and a half ago. Meanwhile we have OGG and even newer MS/Real codecs. I don't see them comparing with the ogg codec wich is considered now the open industry standard. I have made the migration for a really big radio station from Windows Media to ogg, BUT based on a demonstration of the clear qualities of this open codec. You can listen a 22khz, 16 bit, mono stream at 20kbps (more than dial-up friendly). You have CD quality at 64kbps VBR (insignifiant for any broadband connection). All this using ogg. You have support for it in most of the music players around. Why don't I see a relevant competitive analasys between this and aacPlus? Why should I care about it being better than codecs that are mostly irellevant at this moment?

  3. Re:What about other codecs... by kevinadi · · Score: 5, Interesting

    Blame MPEG for creating confusing standard :)

    Anyhow, the MPEG-2 AAC and MPEG-4 AAC are basically identical, except for the addition of some coding tools designed for low bitrate encoding, like internet radio.

    There are some profiles for AAC encoding, which are (in decreasing quality) Main, Low Complexity (which we see in FAAC and Apple's), Low Delay, and the newest is High Efficiency which is low bitrate. There's also a scalable profile thrown in for good measure. I presume AACplus is actually AAC-HE. The technology they're using is from MP3plus we've seen quite some time ago but never takes off. So rest assured that you're not missing anything if you got your collection coded in AAC-LC.

    Also, the previous poster is correct. The psychoacoustics are not defined in the standard. Hell, even the encoder is not actually defined. They only define the decoder and the stream format to ensure interoperability. But yes, obviously MDCT sizes are clearly defined otherwise you can't reverse transform the coefficients. But if you so choose you can ignore their specification on transient handling and your stream will decode correctly, although with crap quality.