Quality setting being equal, from the encoder having to allocate more bits to encode a particular source file, it does not follow that the resulting lossy file is any farther removed from its source than is a file of a lower bitrate from its source. It simply means the encoder judged it required more bits to attain the same (approximate) quality. That’s the entire point of settings based on quality rather than bitrate, after all!
Way off track, though there’s little point going into great depth (as if I even could!) when there’s bound to be plenty of documentation on VBR explaining its actual rationale. But in short: The best lossy encoder in the world would still benefit from a VBR model because all passages of audio are not created equal. It has nothing (necessarily) to do with inherent incompetence of either the encoder or its format as a whole.
Sounds like ABR to me, your aforementioned misconception notwithstanding.
As have the VBR algorithms of all the leading audio codecs. Again, it has nothing to do with intrinsic deficiencies of lossy audio encoding. VBR just is logical given the heterogenous nature of most source material. It makes sense that end-users would prefer oscillating bitrate to disruptive changes in quality.
When I say deficiencies I mean, say for example the encoder can only represent everything sinusoidally, it would not be able to represent a square or triangle wave in terms of sinusoidal waves,
So in a way, VBR tends to be a cheap work around of sorts, as I'm pretty sure a lot of the variation in bitrate could be removed if more flexible algorithms were introduced into the encoder.
When you say relatively few sine waves (surely that depends on the resolution?), isn't that more than only one that would be necessary if it were simply a sine wave? If there were a square wave mode or something it would surely require less bits than choosing to represent it with sine? Perhaps the square wave was the poor example, but I think you yourself may be able to offer some more potent ones.
@Xanikseo: While everything is representable by series of sine waves, only a square wave is representable by a square wave (that is, without applying a filter after that, which makes it more like a sine wave).
@OP: This test shows that 96 kbps QuickTime AAC is superior to Lame -v5, so at 128 kbps it should be much better: http://www.hydrogenaudio.org/forums/index....showtopic=66949You can encode to qtaacenc using foobar: http://www.hydrogenaudio.org/forums/index....showtopic=78072I'm happy with Q65, which actually averages out to about 120 kbps in my collection.Granted, I haven't tried Ogg Vorbis and can't speak for or against its quality, but AAC is a native codec to the iPod, and you won't have to muck around with alternate firmwares, etc. to get vorbis working.(Edit: Whoops, didn't see that you gave your iPod away. Still, AAC is much more widely supported than ogg vorbis...)