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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."

19 of 143 comments (clear)

  1. What about other codecs... by RizwanK · · Score: 3, Interesting

    I was under the impression that the sat broadcasting folks used MP2, optimizing quality and losing some of the psychoacoustic flaws inherent in Layer 3. I last heard about this when I swung by Sirius Radio though, and this was 2001. Anyhow, I'm finally starting to get things coded in AAC, and now theres another subset?!

    1. 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.

  2. low bitstreams not so bad by thehink · · Score: 2, Interesting

    net radio is not bad at all, and this codec looks to take it to the next level. when you're just casually listening, a 56kbps stream does a decent job of giving you what you want to listen to. I find that pretty impressive. i've listened to 56-96 kbps streams, and while not perfect, its virtually as good as analog radio, depending on the music type. anything involving distortion will sound fine. I just find it cool that a low bandwidth stream can successfully push out decent audio content.

  3. 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

  4. Avoiding the 'L' word.... by mAineAc · · Score: 3, Interesting
    If you're on a Mac or other non-Windows computer, install the free VLC player instead of Winamp.

    I like how they avoided using th 'L' word in their report.

    1. Re:Avoiding the 'L' word.... by Anonymous Coward · · Score: 3, Interesting

      BSD? Unix-based? Amiga? BeOS? SkyOS? ReactOS? Hurd? Atheos? Plan 9? VMS?

      Oh, that's right. Linux is the only acceptible non Windows/MacOS operating system.

  5. XM @ 40kbps per music channel, quality still OK by bishr · · Score: 3, Interesting

    I subscribed to XM for about three months, and one of the main reasons I canceled was that the quality was not quite what I wanted. It was pretty good, but some of the "harshness" that you get with lower-bitrate Vorbis, AAC, etc, with cymbals, was pretty jarring to me. I've reencoded files in OGG, WMA at 64kbs, and it's fairly equivalent (though, of course, this is IMHO and therefore totally subjective.) I haven't tried lower bitrates, but as I recall, Vorbis scales downward very well. This may or may not be the new champ for low bitrate sound quality, but this is NOT revolutionary.

    Speaking of XM, it seemd to be feast or famine- either they're playing stuff I like on several channels at once, or I flip around for an entire hourlong drive withouth finding anything - the other main reason why I canceled.

  6. Low bit rates works well with speech. by MtViewGuy · · Score: 3, Interesting

    I think while these low bit rate transmissions might not be great for music, they do work pretty well for transmission of mostly speech broadcasts such as news, radio talk shows and sporting events.

    I think because we're so used to talking over landline telephones with its relatively poor sound quality, Windows Media and Real audio streams transmitted at 16 kilobits per second and the audio stream mentioned in the article sounds reasonably well for mostly-speech programming.

  7. platform irony by BeerCat · · Score: 2, Interesting

    funny, really, that on Windows (where WMA is pushed as the "standard" - even though there are all the other alternatives), Winamp can cope with the new format (superset of AAC), while on the Mac (where AAC is now pushed as the "standard", at least for iTunes / iTMS), it's a bit harder to get a player.

    OK, so Winamp isn't installed by default, but is is becoming the player of choice for the IT cogniscenti in place of WMP, whereas other Mac players are still the curiosity compared to iTunes.

    --
    "She's furniture with a pulse"
    1. Re:platform irony by denniscpearce · · Score: 2, Interesting

      yes, another vote for fb2k.

      its also great as a utility. it handles cue sheets excellently and makes encoding lossy single-file-per-song files from a lossless single-cd-file about as easy as anything. it also takes care of directory structure and tagging very well.

      its an amazing program. i hate having to use winamp. fb2k handles so many audio formats (download the 'special' installer. its probably one of the best piece of software ive ever known. it has lots of support for replaygain (i dont really use it much) and really everything else you could want. also very customizable in the way it looks (if you are into minimalism), but i tend to leave it fairly plain. see some formating strings:http://pelit.koillismaa.fi/fb2k/strings.ph p?s=vote

  8. Good for broadband too by vladd_rom · · Score: 2, Interesting

    My company has around 100 employees, and our net connection is a 1 Mbps line. Needless to say that not all of us can afford a decent 128 kbps streaming.

    This new format is good not only for dial-up but also for broadband corporate connections that seem to die to a crawl when people start using current streaming technologies over them.

  9. I'd settle for peercast working by bofkentucky · · Score: 2, Interesting

    let my listeners spread the bandwidth needed for 64Kb/s OGG streamed by icegenerator/icecast2 amongst themselves, but it will not stay up either on windows or FreeBSD.

    --
    09f911029d74e35bd84156c5635688c0
  10. Re:The Interesting Bit is in the Last Paragraph by DarkMantle · · Score: 2, Interesting
    Interesting...
    At 128 kbps, it can deliver 5.1 channel surround sound.
    See, the funny thing is. Ogg-vorbis supports 5.1, I just can't find an encoder that will use it. And you can encode 5.1 at any bitrate since it uses that bitrate/channel when encoding in more the 2 channel setups.
    By the way, if you know of an ogg encoder that will support 5.1 let me know, I don't want to develop it myself, I don't have time.
    --
    DarkMantle I been bored, so I started a blog.
  11. Re:Just a random thought here, by mattspammail · · Score: 3, Interesting

    Satellite radio is only one way communication. It sounds choppy on the voice channels, because they use a lower quality bitrate. The music channels have a higher bitrate. The number of listeners is not a factor, since it's one way.

    --
    Now accepting PayPal donations!
  12. 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?

  13. Here's how 24kbit/s MP3 sounds (Lessig audiobook) by turnstyle · · Score: 3, Interesting
    When I put the Lessig audiobook together, I finally settled on 24kbit/s MP3s (in true mono).

    Listen to Ch.1 by Doug Kaye and/or Ch.13 by George Sessum, as those files were properly recorded (some of the others were first-time recordings, and they didn't get their levels right).

    --
    Here's what I do: Bitty Browser & Andromeda
  14. Re:The Interesting Bit is in the Last Paragraph by costas · · Score: 2, Interesting

    A few reasons: first of all, it's not just a question of overall bandwidth: maybe you only want to give 64kbps out of your DSL connection to your streaming radio station and let the rest be used by BitTorrent. Second, if you listen internationally to US radio stations, as I do, aacPlus can be buffered more easily at 24kbps unlike MP3 at 128kbps, and because the traffic "weather" between here and the US can get very choppy during peak hours. Third, as the article points out, 24kbps can easily fit into a GPRS/UMTS connection and be streamed over a mobile phone.

  15. Re:The Interesting Bit is in the Last Paragraph by Anonymous Coward · · Score: 1, Interesting

    I got curiuous and looked for it: Ogg vorbis supports up to 255 simultaneous channels (the channels aren't however coupled (yet).

    It's mentionned here at the end of the page:
    http://www.xiph.org/ogg/vorbis/faq.html

    something about the coupling :
    http://www.xiph.org/ogg/vorbis/doc/stereo.html

    You can encode to multichannel from raw audio input it seems/I think (haven't tried it):

    The program "oggenc" has an option "-C" where you can define the number of channels. This is a command-line-tool. It seems it was icnluded in the package "vorbis-tools" in my linux-distribution.

    Hopes this helps some,

    Michel

  16. New low-bitrate champ by Guspaz · · Score: 2, Interesting

    I downloaded the reference source for the AACplus encoder/decoder, and ran a quick test on it.

    At 24kbit, Vorbis needs to encode at 16khz stereo to hit the target bitrate.

    At 24kbit, AACplus can encode at 48khz stereo and still hit the target bitrate.

    Doing a direct comparison, there is no competition at all. 48khz vs 16khz, aacplus wins.

    While I'm very happy that such a huge leap has been made in low-bitrate audio encoding, I'm troubled as to how far Vorbis has fallen behind. They don't seem to have made any major improvements in audio quality in years.