[vorbis] When will quality increase be unnoticable?

David Tenser david.tenser at telia.com
Thu Jun 20 13:30:21 PDT 2002



I agree with much of what you say, Øyvind, but I have to ask you a 
question, and I hope that you can be perfectly honest: Can you really 
tell the difference between quality 7 and 8? If you can, can you give me 
an example of a good song that it's fairly easy to hear the difference? 
If not, why not enqode all your music in quality 7 instead?

/ David Tenser

Øyvind Stegard wrote:

> 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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