is Ogg Vorbis the best Lossy encoder? (what is?) Options

[snadge]

ive been trying to find out which encoder produces the best audio quality at similar bitrates (say 128k) to other lossy encoders - ive seen a couple of reviews and tests and they seem to point to Ogg Vorbis as being the better encoder - I encoded a WAV file to MP3, AAC, WMA & Vorbis all at 128k CBR and while it was hard to tell any difference the Ogg Vorbis was pumping out higher frequencies at 20Khz + and the rest were up to 18Khz (according to spectrum on Foobar2000)

the only gripe is during conversion Ogg Vorbis was the only one that didnt have the Tag info from the WAV file.

can anyone lead me to more listening test results and the likes so i can ascertain for myself? - i would like others views on "best lossy codec" etc

I read on Wikipedia that AAC at 128k is supposedly Transparent to the lossless source according to listening tests by M-PEG for ITU or something? - I was hoping to adopt AAC as my default rip codec as I have an iPod and MP3 player that plays AAC - but Iam in the market for a new media-player so all this is irrelevant as I can buy something to suite.

thanks in advance for replies

please be gentle...im a noob at all this,

This post has been edited by snadge: Jun 8 2011, 04:17

[Xanikseo]

I never said that you should use a square waves as the basis function, but only when it would be appropriate, ie when it would reduce the bitrate. Unfortunately, square waves were a bad example, as they do not occur so often in music, but the idea is there - that there are samples that are very hard to approximate using sine waves. Yes there are many advantages to using sine waves, since it is an area of Mathematics that is very well covered. Perhaps we are starting to reach the boundaries in that area, and it is time to explore other ways of approximating signals which should be used when sine waves would be less effective.

By the way, it is not possible to produce a sine wave with square wave samples, as you would need an infinite amount of them to produce a perfectly smooth sine wave curve. The only reason sine waves are used in audio compression is because it is (mostly correctly) assumed that the smoothness of the sine curve was lost when the sample rate decreased from an infinitely high frequency to a meagre 44.1KHz (for example). Similarly, you can only reproduce a perfect square wave with an infinite sum of sine wave, which is why I suggested a square wave mode, but apparently square waves are not a common occurrence in music.

According to the most recent listening test for low bitrates, CELT performed the best. IIRC this is due to the fact that as little of the bitstream as possible is used for defining things like switching block sizes / encoding modes etc. This fits with what you were saying, Jaz, that the lower overhead is of an advantage at very low bitrates. Apparently this also causes the bitrate to be a sort-of restrained VBR, in that the bitrate does not jump suddenly, but changes gradually. It is interesting that CELT does so well, considering the amount of compromises that were involved in the design of the codec.

However I still feel more flexibility would result in lower overall bitrates in certain circumstances and less needs for jumps in bitrate. Perhaps a more flexible way of informing the decoder when to activate and deactivate each "encoding mode" is needed (are these sorts of things CBR?, as in, is there always the same amount of information given to the decoder per second about this? Perhaps this should also be VBR-ised. If one "encoding mode" was used for 10 seconds without changing, it would be pointless to keep telling the encoder every 100th of a second that it should keep using the same "encoding mode"!). This will probably result in a higher memory usage though...

This post has been edited by Xanikseo: Jun 15 2011, 23:26