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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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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?!
But hey, what do I know?
How we know is more important than what we know.
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.
I'm not an ogg-head but I was pleasantly surprised by the quality of 32 Kbit ogg streams a while ago.
g .html
http://www.virginradio.co.uk/thestation/listen/og
I like how they avoided using th 'L' word in their report.
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.
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.
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.
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"
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.
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...
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
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.
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).
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The one thing which will revolutionize Internet radio (and Internet TV and filesharing) is IPv6 with working multicasting. No longer do you need a fat pipe to service hundreds or thousands of listeners. You can run a popular radio station over your DSL line if you want. AAC and other codecs are just babysteps which are immediately undone by licensing and DRM issues.
aacPlus is just a marketing name for the HE-AAC standard.
There are GPL'ed implementations of HE-AAC decoders, for example at http://www.audiocoding.com, so these streams should be playable on open source systems, too.
Btw. Some of technical details in the article (notably about parametric stereo) are *complete bollocks*. What they describe is Mide-Side stereo.
Parametric stereo transmits only a mono channel plus a very small amount of sideband information that describes how to reconstruct the stereo image (via decorrelation and fading).
Sorry, but I have to say mp3 streaming is crappy. Just because most players support it, doesn't make it good.
AAC is indeed better.
I just wish the general public would download newer players that supported things like Vorbis, AAC.
But unfortunately,
mp3 = music file
Not "format of music file". but "music file". If it's not mp3, it's not a music file.
I think step 1 is to get rid of this carma that mp3=audio. make mp3=old audo format.
Until we do that... mp3 will be sticking around, and sucking.
Yes, HE AAC and AAC+ are the same thing. HE AAC is the name that MPEG gives it, and AAC+ is Coding Technologies name for their implementation.
Next up is AAC PS, for parametric stereo, which applies the SBR techniques to synthesizing stereo. Gives another big leap yet for music listening - 24 Kbps is good enough for people who can live with MP3 @ 160 or so.
My video compression blog
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.
Some people wind up saving mono files that duplicate the audio on both right and left channels, rather than save it with a single mono channel.
You wind up with a file that's twice as big, with no benefit.
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