conversion between FAAC, Lame MP3, AoTuV Ogg Vorbis, WMA and RA

[MetalheadGautham]

Among the above formats, what is the loss of quality during conversion?

I want to know how much of quality will be lost if I make the following conversions(at the same bitrate, please say if a lower one will do). I heard that transcoading between lossy formats causes reduction in clarity, so:


1. from Lame mp3 to OGG Vorbis

2. from ogg/mp3 to wma

3. from AAC to wma

4. from AAC to mp3



and please tell me a format to which I can convert RealAudio to so that there is minimal loss in quality and maximum reduction in size(32 kbps real audio)

[foxyshadis]

Quality of the input determines the total amount of loss; you can't predict quality based on its filetype whatsoever. All you can do is make vague guesses or actually listen. What's the point of converting from one format to another at the same bitrate anyway?

Current WMA and Vorbis are similar and superior to MP3 under 112kbps, and have a very marginal advantage above that. AAC LC is generally marginally superior to MP3 at high and low bitrates. AAC+ (HE) is superior to MP3 under 112, and very superior to all other codecs under 64, but sounds worse over 112. (Bitrates may vary based on source complexity, these are rough guides only.)

I have no experience with any recent realaudio codec, and HA doesn't seem to care either. But you want a reduction from 32kbps? Good luck, even AAC HE/PS won't make that sound good. Forget reduction and concentrate on standardizing, even if it ends up a little larger, it'll have to be to keep any quality.

[MetalheadGautham]

the reasons are as follows:

1. my mp3 player supports only wma, mp3

hence i need to convert my ogg, aac, ra files to one of the above.


2. I like foobar2000, but it cant play RA, and it isn't much supported even otherwise. I have tonnes of 32, 20 kbps RealAudio music, I also have some 48, 20 kbps cook music. I want a format(s) and bitrate to which I can convert them to.

3. I want to unify my music to a single format.

[Dark Shikari]

32kbps, 20kbps RealAudio "music"? At that kind of bitrate it would hardly be music.

Sounds like you've done a lot of streamripping at really really bad internet radio stations.

[foxyshadis]

You have to pick one: Even lower quality, much larger file, or can't play it anywhere. Go with the second, convert it to 64 or 80kbps mp3 (or wma), and enjoy. There's no realistic way to prevet them all, and the added complication of not having aac or vorbis on the DAP cuts your choices down a lot.

[MetalheadGautham]

32kbps, 20kbps RealAudio "music"? At that kind of bitrate it would hardly be music.

Sounds like you've done a lot of streamripping at really really bad internet radio stations.

nah... For radio I rip with audacity(aka vorbis)

I got mine from .ram files using flashget:cool:

[MetalheadGautham]

Now for a MAJOR doubt:(please tell me if my conclution is right)

Facts:

lame is opensource

vorbis is opensource

lame uses advanced psychoaccoustics

vorbis also uses advanced psychoaccoustics

lame came before vorbis, and is being developed to the limits of MP3(which is being expanded; VBR, 512 kbps, etc are realities)

vorbis must borrow a lot of stuff in the removing unnessary data part from lame, as both are opensource projects(and really good ones to)

hence the main differences between files of the same QUALITY between vorbis and lame is the SIZE

also, vorbis performs much better at lower bitrates.

then shouldn't it be really easy to convert mp3 to vorbis without quality loss at the same time reducing size?

[smok3]

facts:
- transcoding is evil
- mp3 is the most supported format (so lame)
- real files can be he-acc so they can at least theoreticaly sound half-decent at 32 kbps
- do your own blind tests (abx, abchr...)
- transcoding is evil

yeah, its hard..

[MetalheadGautham]

I ALWAYS use a wav file as an intermediate file

Thanks! even I thought that he-aac is the best thing to convert a real audio to. I heard a lot of its properties of being good at 32kbps

but my conclusions are still the same - it must be easy to convert mp3>wav>ogg without trouble....


can anyone point me to a very reacent listening test so that I can decide some things?

[foxyshadis]

Presented logically like that it sounds good, but the mathematics just don't work; the quantization methods, spectral bands, and almost every other implemenation detail of each standard are totally different. Only mp3 and aac share anything beyond basic principles, and they're still too different to have a lossless transfer like that.

[smok3]

I ALWAYS use a wav file as an intermediate file
thats still called transcoding.