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Music Media The Internet

Low-bandwidth Net Radio 143

Posted by michael 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."
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Low-bandwidth Net Radio More

Low-bandwidth Net Radio

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  • What about other codecs... (Score:3, Interesting)

    by RizwanK ( 813432 ) <rizwank@geekyDEB ... com minus distro> on Saturday January 22, 2005 @08:08AM (#11440686) Homepage
    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?!

    • Re:What about other codecs... (Score:5, Insightful)

      by tomstdenis ( 446163 ) <tomstdenis.gmail@com> on Saturday January 22, 2005 @08:11AM (#11440696) Homepage
      Um... psychoacoustic modelling IIRC isn't part of the standard. The standard mandates things like bit format and DCT precision.

      So if your MP3s sound like crap

      - up the bitrate to something reasonable
      - Get a good source to encode from
      - change the encoder [lame -q 0 is great]

      Tom
      • Um... psychoacoustic modelling IIRC isn't part of the standard. The standard mandates things like bit format and DCT precision.

        Well, strictly speaking any subband coder uses psychoacoustical modelling since it depends on frequency masking, but IIRC layer 2 and layer 3 both use subband coding.
      • Or for the best that Lame 3.90.3 can put out, use the --alt-preset-* presets. I've been very happy with --alt-preset-standard. And yes, I intentionally use 3.90.3 and not the newer releases.

    • Re:What about other codecs... (Score:1, Funny)

      by Anonymous Coward
      Anyhow, I'm finally starting to get things coded in AAC, and now theres another subset?!
      Yeah, it's Murphy's Law. It states if you compress something enough you are a Nazi, oh wait...

    • Here's how 24kbit/s MP3 sounds (Lessig audiobook) (Score:4, Interesting)

      by turnstyle ( 588788 ) on Saturday January 22, 2005 @09:01AM (#11440883) Homepage
      When I put the Lessig audiobook [turnstyle.org] 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).

    • Re:What about other codecs... (Score:5, Interesting)

      by kevinadi ( 191992 ) on Saturday January 22, 2005 @09:51AM (#11441045)
      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.
      • >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.

        Well, HE-AAC got accepted into pretty much every broadcasting standard there is. I don't think you can "take off" more than that. Customers generally aren't aware of it, but that doesn't matter - companies sell solutions (iPod) rather than formats (AAC) anyway.

        BTW. Y
        • Nah, what I meant is MP3plus (or is it MP3pro?) never takes off. The technology in MP3pro is actually pretty inventive for low bitrate, MPEG decided to use it for AAC-HE.

          Anyway it's all for the best. We get better quality music that's actually decent using dial up with a format no one company controls so we can use our player of choice. I noticed the steady decline of realaudio content that requires a pain in the ass player to listen to.

      • HE AAC==AAC+ (Score:3, Informative)

        by benwaggoner ( 513209 )
        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.

  • Just a random thought here, (Score:5, Funny)

    by QuantumG ( 50515 ) <qg@biodome.org> on Saturday January 22, 2005 @08:09AM (#11440690) Homepage Journal
    but maybe the other stations are choppy because there's actually a large number of people listening to them at the same time.

    But hey, what do I know?

    • 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.
      • Being one way communication does not mean it does not use bandwidth. Even most internet radio stations use one way UDP data streams of information and still use alot of bandwidth. I think the word you are looking for is broadcast.

        • Re:Just a random thought here, (Score:4, Informative)

          by ZorinLynx ( 31751 ) on Saturday January 22, 2005 @10:36AM (#11441228) Homepage
          Satellite radio is a broadcast medium, which means one signal is sent down to a large area, and anyone in that area can receive the same signal without quality loss as the number of listeners goes up.

          It can be compared to any other radio broadcast; just because you're listening to 99.9 RIAA-0wn5-j00 FM doesn't mean other people have a weaker signal or diminished sound quality.

          -Z
          • just because you're listening to 99.9 RIAA-0wn5-j00 FM doesn't mean other people have a weaker signal or diminished sound quality.

            they wont have a diminished sound quality (for the most part, if its all on the edge of the range they might). but they most definately will always have a weaker signal in the immediate area. this is because your antenna itself distorts the field around it when it attracts the singal, and a small amount of energy is used in the reproduction of the sound wave when the receiver

            • Re:Just a random thought here, (Score:1, Informative)

              by Anonymous Coward
              Yes, such a low drop as to even not be worth discussing.
              • I used to be an announcer at an AM station and kept the transmitter log during my shift as well as the program log, which meant that I recorded plate voltage and plate current for the transmitter at regular intervals. Since the transmitter was in a room adjoining the studio I looked in on it several times per hour in addition to the "official" observation times. Maybe it was just atmospheric changes but it seemed that what I saw tended to reflect a varying "load", i.e., number of tuner front ends adjusted
                • Definitely weather conditions, or varying power conditions (voltages sometiems swing up and down during the day as load changes)

                  A receiver tuned into a radio broadcast may affect the signal as it passes it, but it won't affect the transmitter in any way.
  • 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.

  • Ogg streaming seems pretty good (Score:5, Interesting)

    by Xenna ( 37238 ) on Saturday January 22, 2005 @08:17AM (#11440719)
    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 [virginradio.co.uk]
    • Dude, you said ogg in a discussion about audio encoding on slashdot. I envy the size of your karma.
    • I also think ogg vorbis is surprisingly good at low bitrates. I serve 64 kbps radio streams for myself, which are stereo and sound better than FM radio with respect to frequency response -- both bass and high freq remain, and it is very pleasant to listen to unlike low bitrate mp3 or wma. I think low bitrate ogg vorbis is underappreciated!
      • Vorbis is great at low bitrates.

        But after listening to a 24kbit stream of AACplus, I have come to the conclusion that Vorbis just got it's ass handed to it at low bitrates. Seriously, 44khz stereo at 24kbit and it sounds great.

        I'm trying to find an AACplus encoder somewhere to do some side-by-side comparisons.
    • Parent is absolutely on the money. Our college's radio station, though operated independently of the institution, isn't exactly the most tech-savvy group of people, but when they began webcasting their stream again (after that whole pay-per-listener thing was waived for certain nonprofits) someone must have shown them the quality difference between mp3 and ogg, b/c they're streaming with ogg now at 67kbps. Give a listen if you're interested: http://engine.collegemedia.vt.edu:8000/wuvt.ogg

      It's the weeken
    • Us too. But we're interested in AACPlus as well; it seems worthwhile looking into.

      James (who works at Virgin Radio)

  • Avoiding the 'L' word.... (Score:3, Interesting)

    by mAineAc ( 580334 ) <mAineAc_____ AT hotmail DOT com> on Saturday January 22, 2005 @08:20AM (#11440728) Homepage
    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.

    • Re:Avoiding the 'L' word.... (Score:3, Interesting)

      by Anonymous Coward
      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.
      • Not the only, but I would say the most widely used non-windows/non-mac OS.

      • Re:Avoiding the 'L' word.... (Score:1, Insightful)

        by Anonymous Coward
        I seriously doubt that you can run VLC on Amiga, ReactOS, Hurd, Atheos, Plan9 or VMS. But If you're hiding ports on these systems somewhere, please share with us.
        Besides, why make MacOS a special case? The cumuled marketshare of all Linux distros is well over the Apple one.
        Please, don't try to hide the bias when it's obvious.

        Ah, the joy of posting anonymous.

        • Besides, why make MacOS a special case? The cumuled marketshare of all Linux distros is well over the Apple one.

          Disclaimer: I an not trying validate the ignoring of Linux. I use quite often myself.

          I think it's not as much a matter of how many overall users there are as it is who they are. While there are lots of Linux installs out there, how many are actually desktop systems rather than servers? And of those, how many are someone's primary system? Etc.

          In terms of mainstream users, Linux still seem

  • XM @ 40kbps per music channel, quality still OK (Score:3, Interesting)

    by bishr ( 262019 ) on Saturday January 22, 2005 @08:21AM (#11440734)
    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.

    • Re:XM @ 40kbps per music channel, quality still OK (Score:1, Informative)

      by Anonymous Coward

      I've reencoded files in OGG, WMA at 64kbs, and it's fairly equivalent

      You're reencoding from XM and trying to compare quality? The XM codec has already thrown away lots of information, transcoding to another format is only going to throw away more, it's certainly not going to magically get the information back somehow.

      It's like chopping an apple in half, and trying to determine whether you can chop one of the halves in a way that gets you more than half an apple. Impossible by definition.

  • The Interesting Bit is in the Last Paragraph (Score:5, Insightful)

    by conJunk ( 779958 ) on Saturday January 22, 2005 @08:24AM (#11440744)
    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.
    • 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.
      • Develop it, you'd be cool for doing it.
        • Develop it, you'd be cool for doing it.

          I wish open source advocates would quit saying stuff to the effect of "write it yourself". Even though it probably isn't meant to be insulting, not a whole lot of people can actually do it and do a good job of it. Do a bad job and it's probably easier for contributor to rewrite it from scratch than it is to advance the project.

      • Ogg-vorbis supports 5.1, I just can't find an encoder that will use it.

        What's the problem? oggenc will have no problem encoding from a 6-channel wav file.

        The problem right now is that the ogg encoder doesn't do the differential-encoding thing, saving only the differences between each of the channels, making the stream much smaller. It's capable of doing it, but the code hasn't been written yet.

        As a matter of fact, Ogg/Vorbis in general hasn't really been updated since it's 1.0 release years ago. Someb

    • 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.

      So? That doesn't mean that server load and server bandwidth isn't a factor preventing people from getting into the game, reducing these two means more people can use a server.
    • 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 GP

    • Re:The Interesting Bit is in the Last Paragraph (Score:1, Interesting)

      by Anonymous Coward
      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

    • Truth is, everyone (at least in the west and industrialized Asia) has or will get broadband

      Wired broadband and fixed wireless broadband do not count. It has to be a mobile connection, or it won't stand a chance of replacing Clear Channel's FM and XM programming in motor vehicles. Currently, affordable mobile connections are rather low-throughput, so they'll need a decent codec.

    • There are those of us who don't have and can't get broadband, no matter how much we are willing to pay. Those who think broadband is pervasive have most likely not visited the large stretches of rural America where you're lucky to be able to get 33.6K over copper phone lines. Of course, we're the target market for non-Net XM and Sirius :)
    • I personally think you are thinking about this from the wrong end. In ten years we'll all have enough bandwidth to reliably stream multichannel sound ripped right off a DVD and not recoded or transcoded. The big advantage is the quality at low bitrates. Why? Cellular. The next big thing (tm) is going to be the proliferation of the mobile internet that we've been hearing about for so long. If you have a good signal you can get over 24kBps even with GPRS. Newer standards allow quite a bit more than that. You
    • Consider this: maybe I don't care about 5.1 surround, but I *still* want higher compression. Why? Right now, I basically have to dedicate my internet connection to listening to streaming audio. I would *love* for streaming audio to be low enough bit rate that I could, for example, play online games while listening to streaming audio in the background. Or, possibly as low-bitrate high-quality technologies advance, and simultaneously, broadband becomes more prevalent and higher speed, you could see all sorts

  • Low bit rates works well with speech. (Score:4, Interesting)

    by MtViewGuy ( 197597 ) on Saturday January 22, 2005 @08:27AM (#11440753)
    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.
    • That's a good point. I'd even go farther and say we can understand human voice even when the quality is bad because it's hard wired in. Deciphering a guitar or drum from a song isn't and so we need higher quality since we have to think about it more.

      I'm no scientist, I just like to observe things and try to come up with a reason why they happen.
      • That's almost true. Actually human _ear_ is geared toward speech. Our ears are much more sensitive at the frequencies of human speech, and much less sensitive at higher or lower frequencies. This is the reason why MP3 and AAC can sound good (and to a greater extent, DTS and Dolby AC-3 which are used in the cinemas). They encode low and high frequencies with less accuracy, because we don't perceive those frequencies as well as we should.

        The exact point you're making is called the fields of psychoacoustics,
      • "I'm no scientist, I just like to observe things and try to come up with a reason why they happen."

        Allow me to suggest this as your sig file.

    • Actually the purpose of those technologies are specifically geared toward music. For speech, there are many researches done for exactly that purpose. The state of the art in speech coding can go as low as 4 or even 2 kbps AFAIK while maintaining toll quality speech.

      Your ordinary GSM cell phone works at 16 kbps, off the top of my head, I don't exactly remember. Your landline works at roughly the same bitrate. The reason why we don't see an increase in speech quality is due to existing equipment that'll be t
      • Audio coders can have the advantage of simplicity. For example, delta modulation, which is easy to implement, has very low latency and degrades gracefully on high BER channels.
        • True, especially considering most audio coders are based on transforms rather than linear prediction.

          The overkill part I'm referring to is the audio specific psychoacoustic processing which requires even heavier calculation than linear prediction since it has to calculate many variables. But as soon as it leaves that stage, everything else is quite simple by comparison.

          But then again, when working at very low bitrates and the application is speech specific, audio coders simply can't compete with speech co

  • platform irony (Score:3, Interesting)

    by BeerCat ( 685972 ) on Saturday January 22, 2005 @08:28AM (#11440758) Homepage
    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.

    • 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.

      Last I heard, WinAmp was a discontinued product. That's irony for you.

      The article mentions that VLC can play AAC+. I bet VLC is installed on most desktop Macs used by the "IT cogniscenti", alongside iTunes. Furthermore, iTunes supports audio format plug-ins (I don't know whether there exists an AAC+ plug-in.)

    • Re:platform irony (Score:4, Insightful)

      by moonbender ( 547943 ) <moonbender@@@gmail...com> on Saturday January 22, 2005 @10:43AM (#11441257)
      OK, so Winamp isn't installed by default, but is is becoming the player of choice for the IT cogniscenti in place of WMP.

      Hm. First off, I wouldn't say that Winamp is becoming anything - it already is, and has been for a while. People, and not only "IT cogniscenti" (aka geeks), have been using Winamp in the days when WMP wasn't a generally known acronym. To me, Winamp was the player of the period when MP3 was still new (remember oth.net and AudioGalaxy?). I kind of doubt the number of users is still increasing, in fact I imagine that if anything, the number is decreasing.
      I might be wrong, though - so, what is the choice among the geeks these days? Do you all still use Winamp? Personally, I've been using Foobar [foobar2000.org] for a long time now, mostly because of it's small footprint, straightforward interface and out-of-the-box global hotkeys. Because I'm so happy with it, I really haven't even looked out for any other new players, so I'm curious as to whether I've missed anything. (And I don't mean iTunes for Windows.)
      • 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
      • I'll have to echo this myself. I remember using Winamp in the days of Napster when it was just a baby (both were). It was the best in the day and is still the best (for audio). For video I normally use Media Player Classic [sourceforge.net].
  • 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.

  • It's good to see (Score:5, Insightful)

    by Dorsai65 ( 804760 ) <[dkmerriman] [at] [gmail.com]> on Saturday January 22, 2005 @08:30AM (#11440764) Homepage Journal

    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.

    • 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.

      Unless you're streaming mostly music, you really don't need the highest quality data transmission rate for streaming audio over the Internet. Run Real or Windows Media audio streams at 16 kbps and the sound quality is more than acceptable enough to hear mostly speech broadcasts such as news, sporting events and talk radio clearly.

  • I'd settle for peercast working (Score:3, Interesting)

    by bofkentucky ( 555107 ) <bofkentucky@NOSpAm.gmail.com> on Saturday January 22, 2005 @08:30AM (#11440765) Homepage Journal
    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.

  • i dont get it (Score:5, Interesting)

    by hasst ( 852296 ) on Saturday January 22, 2005 @08:42AM (#11440806)
    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?
    • See for example:

      http://www.rjamorim.com/test/32kbps/results.html

      Vorbis is simply not competitive to HE-AAC at such low bitrates.

    • Re:i dont get it (Score:3, Informative)

      by benwaggoner ( 513209 )
      I think you're missing the point. HE AAC is more than twice as efficient as today's leading class of codecs (AAC, WMA, Ogg). Twice is a big deal! Think of the difference between, say, MPEG-2 and H.264 or WMV9 Advanced Profile. It took video codecs a full DECADE to get the kind of improvement jump we're getting with HE AAC. That 20 Kbps stream can be a great sounding 44.1 with HE AAC - better than that 64 Kbps VBR stream you cite.

      The technology has been around for a while in enterprise systems, but is only
    • There is no "Ogg codec". Ogg is the encapsulation format (think AVI or MPEG), Vorbis is the audio codec (think MP3 or WMA). Saying "Ogg codec" is pretty much the same as saying "AVI codec".
    • "You have CD quality at 64kbps VBR .. using ogg"

      If that's the case, you either have crappy hearing, crappy speakers/headphones/amp/soundcard, or some very easy to encode CD's.

  • SomaFM (Score:5, Informative)

    by HoneyBunchesOfGoats ( 619017 ) on Saturday January 22, 2005 @08:59AM (#11440877)
    SomaFM [somafm.com], 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.

  • Revolution? (Score:2, Insightful)

    by Anonymous Coward
    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.
    • >AAC and other codecs are just babysteps which are
      >immediately undone by licensing and DRM issues.

      AAC has no "licensing issues" in this context (no per broadcast fees, unlike MP3".

      Neither does AAC have DRM - this is always added through nonstandard extensions. But you can do that with any format.
    • What is really needed is for the ISPs to support SSM [faqs.org] (Source Specific Multicast). This would allow anybody to stream audio or video in an efficient way. The bad news is that few ISPs have it turned on. The core backbone is enabled, so that isn't an issue. Why isn't it turned on? No demand!

      Call/email your local ISP and tell them that you want SSM support. If enough people call, then they will turn it on (they already have all the equipment). Once turned on, I predict that there will be a flowering of softw

  • aacPlus == HE-AAC (Score:5, Informative)

    by Skuto ( 171945 ) on Saturday January 22, 2005 @09:54AM (#11441055) Homepage
    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).

    • 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).

      so in other words, they transmit mono (L+R, let's call it A) plus information that can be used to reconstruct the stereo signals (L-R, let's call it B, likely to be quite small when L

      • I don't know where you got the idea that the sideband information is "L-R". It's a parameter set for a filterbank.

        Neither is the mono channel (necessarily) L+R.

        The reconstruction isn't *anything* like you describe.

  • Can it get worse than mp3 (Score:3, Insightful)

    by digitalgimpus ( 468277 ) on Saturday January 22, 2005 @10:53AM (#11441306) Homepage
    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.
    • Really you want something lossless if you don't want it to suck... compression always has an effect.

      AAC is DRM'd so I avoid it like the plague anyway.
      • AAC is DRM'd so I avoid it like the plague anyway.

        Huh? You can add a DRM wrapper to AAC files (which is what Apple does in their iTunes Music Store). But regular AAC files (such as the ones you are getting when ripping your CDs with iTunes) are just as 'free' or 'non-free' as regular MP3 files: you can copy and play them on as many devices and computers as you want.
      • I'm not a big fan of AAC, mind you, but it most certainly is not DRM'd. It's a simple audio codec, not much different (in that regard) to MP3 or Vorbis.

        Apple's music store uses a DRM'd MPEG-4 wrapper for their files, which are encoded using AAC. But that means nothing: a wrapper is quite a different thing than a codec. Look at XviD: it's a video codec which can be encapsulated in any number of wrappers (AVI, OGM, Matroska). Heck, you can even stuff an MP3 file into a wrapper and use it that way (Matroska a
  • Well, that's nice for listeners, but if aacPlus is as good as touted, then the real benefit will be to very small indie operators who want to serve up a few streams of their own over a DSL line - more listeners.

    Speaking of which, does Shoutcast or any of the other popular streaming media software packages support aacPlus?
  • But when I tried to listen to one of the 24kbps stations, the crappy quality was very noticable (it was playing My Immortal by Evanescence however, so no great loss, but the highs in the song were very crackly). However 40kbps was perfectly fine. I didn't try one of the 32kbps stations however.

    The 48kbps stations are pretty good quality. I haven't heard a pop or crackle.

    Still, now you 28.8k backwater people can at least listen to net radio that isn't awful.

    Shame Apple didn't use AACPlus in the iPod Shuff
    • It's probably very difficult for Apple to change their system at this point: they could include HE-AAC/AACplus support in the iPod Shuffle, sure, but the users who aren't bringing their already ripped MP3/Vorbis/AAC/whatever collection to the table are going to be presumably purchasing files from the iTunes Music Store.

      Now, if Apple decided to upgrade iTMS to provide AACplus files, then they'd be breaking compatibility with existing iPod models, which is probably not a wise idea, all things considered.
  • What happens when someone hacks their XM receiver (or ham radio) to extract the raw aacPlus data, then streams from a Shoutcast server?
    • What happens when someone hacks their XM receiver (or ham radio) to extract the raw aacPlus data, then streams from a Shoutcast server?

      Then XM sues them and they go to jail for violating the DMCA.

      Next question?
      • OK, what happens when they're a Pakistani tribal chief's brat, who runs a series of StreamTorrent nodes around the Net?
        • OK, what happens when they're a Pakistani tribal chief's brat, who runs a series of StreamTorrent nodes around the Net?

          We "liberate" their country through force ;)
          • Exactly: the "brat" in question is Osama, and blowing up the WTC etc wasn't enough to "liberate" his tribe's 40 acres.
    • ummm you'd get sued, but you could this w/o any hacking.. simple lineout to soundcard.
  • The real revolution will come when it is possible to multicast these type of streams. Today, if 1,000 people tune it, 1,000 copies of the audio data must be transmitted. With multicasting, almost anyone can run their own radio or TV station without having to pay for enormous amounts of bandwidth. Multicasting isn't possible today because not all routers are configured for it, even though IPv4 supports it. I've assumed for a while that when the Internet migrates to IPv6, multicasting will be a goal of that
  • I pay monthly to subscribe to Digitally Imported [digitallyimported.com] Radio. I was a gold subscriber for a year. Then I tried a 2 day platinum trial account and was sold instantly. The Plarinum gives you a 160K stream and it sounds simply amazing. Compressed audio is fine for simple listening, but sounds terrible the louder you turn it up. Even at 128K. All the MP3s I make are at least 192K.

  • New low-bitrate champ (Score:3, Interesting)

    by Guspaz ( 556486 ) on Saturday January 22, 2005 @03:22PM (#11443238)
    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.
  • internet radio streaming is cool, if you don't have any listeners or plans to get them.

    even if you can get decent sound down at 24kbps thats still an extra 24kbps you have to add for every simultaneous listener.

    podcasting's the way to go if you want to do your own audio broadcasts.

    tie it in with blogtorrent and you're good to go.

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