[vorbis] When will quality increase be unnoticable?

[Øyvind Stegard]

Phaedras at gmx.net wrote:

>I started thinking about this after doing a little testing with AAC, MP3 and
>Ogg Vorbis. I was comparing the different formats at similiar bitrates.
>After a while I finally realized that they all sound more or less the same to me.
>>From 160 kbps on, I usually cannot detect any difference between a lossy
>encoding and the original source. If LAME is used, I have to strain to notice
>anything at 128. Ogg Vorbis is the same at slightly lower bitrates.
>
>I know there are enough people out there who, even without ABXing, could
>tell me exactly what sounds different, but personally I have a hard time to
>discern any changes made by a lossy encoder (at a decent bitrate) to a piece of
>music. I also believe the majority of people feel this way. And so I wonder:
>When will codec development stop concentrating on quality and start
>concentrating on size? When I think about where Ogg Vorbis is right now and where it
>will be at 1.0, I don't understand why one would even need these "discrete
>wavelets" that are being discussed. Artifacts, like those that occur with MP3
>seem nonexistant with Vorbis. I don't notice any high-frequency "squishyness"
>How much more of a quality gain can even be achieved?
>
>  
>
Concerning 'Quality vs Size':

1. I don't really think there should be much more decrease in size, 
because, after all, no matter what kind of technology you use or how 
advanced audio coding algorithms you use, one needs to face the fact 
that lots of the original audio information is removed, and can never be 
restored 100%. Too much at 128kbps, if you ask me, and this is, for me, 
the absolute lowest acceptable bitrate (think of how much is really 
removed: 128 reduces the data amount to 1/12th of the original size), 
and audio quality is, no matter which codec (though ogg beats'em all) 
rather poor below 128 kbps. (This is easy to hear if you have decent 
sound equipment/a decent stereo with OK speakers and, of course, the 
sound card in the computer has a lot to do with the final quality.)

2. Harddisks (and storage in general) aren't exactly getting any smaller 
these days, and the same with internet/network bandwith. So why the need 
to reduce SO much, losing audio quality on the way. I think compression 
ratios will go the other way in the future, to preserve more of the 
original sound.

I think that audio codec developers should concentrate on getting more 
quality out of the standard ratios we have today, rather than trying to 
get the bitrate even lower.

I would also like to point out that I am talking about encoding music 
here, not speech (or perhaps streaming of music...) or anything else 
where quality is not the number one priority.

To me, quality is very important concerning music, and therefore I 
encode all my music in OGG format with an oggenc quality level of 8 
(~256kbps). Some might call me a quality freak, but this is still my 
subjective opinion, comments are very welcome. I am quite interested in 
digital audio, though I have no background developing such things, but I 
know the general principles.

Regards,
Øyvind Stegard

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