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Low-bandwidth Net Radio
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."
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Reading the article, my first thought was "so what? So we can ultracompress audio so it sounds good at low bandwidth? What's the point?" Truth is, everyone (at least in the west and industrialized Asia) has or will get broadband, *especially* those who are interested in things like net radio.
Then you get to this bit:
It seems crazy until you try it, but Mostly Classical proves that aacPlus can sound great at 24 kpbs. At 48 kbps, it's almost as crisp as a CD. At 128 kbps, it can deliver 5.1 channel surround sound.
Using the compression to deliver multichannel surround sound is pretty cool. In 5, 10 years, we'll probably have a really flash standard for home audio, and it's nice to know that some folks are thinking ahead to make sure we'll be able to get it streaming on our DSL lines.
that folks are (again) distinguishing between the quality needed for casual use (having background noise) and sit-and-listen-to-it quality (CD/live).
One of my peeves about broadcasting over the net is that so many people want perfect signal, regardless of what they're using the broadcast for. The added bandwidth needed for studio-quality everything just means ever fatter pipes are demanded, raising the cost/price of the whole infrastructure and adding to the net congestion.
--- Asking inconvenient questions for over 30 years...
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?
SomaFM, an entirely listener-supported Internet radio site, has a few streams in aacPlus. I recommend them, they play stuff that you normally don't run across.
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.