Hello all,
this is a forward of a thread on
http://bboard.mp3.com/mp3/ubb/Forum1/HTML/003127.html :
OK, I gave in and did a quick resampling test.
It shows why you shouldn't use lame to do your resampling.
Here's three plots. 1) Original signal. 2) Resampled via Cool Edit, 3)
Resampled via lame.
http://privatewww.essex.ac.uk/~djmrob/mp3board/orig.gif
This is my original test signal. You're looking at a spectral
(frequency domain) plot. The time axis is samples.
There's an upward tone sweep, some very quiet tones, then a downwards
tone sweep with an 18kHz tone mixed in with it (the crossing green
lines on the right).
http://privatewww.essex.ac.uk/~djmrob/mp3board/cep.gif
This is the test signal resampled to 32kHz by Cool Edit Pro, using the
highest quality setting (999). Basically, you have what you had
before, minus the stuff that was over 16kHz (which can't be
represented in a 32kHz sampled system - Nyquist theorem). The
background noise is slightly higher than I'd like (-84dB rather than
-96dB - possible in CEP if you go into the 32-bit domain to do the
resampling, then dither back to 16-bits), but overall, it's a pretty
good resample.
http://privatewww.essex.ac.uk/~djmrob/mp3board/lame.gif
This is the result of the same resampling process, carried out by
lame. As you can't just resample in lame, I used:
lame --resample 32 -h -b 256 orig.wav lame.mp3
then decoded the mp3 using Winamp 2.22. You get the same thing
encoding at 320kbps too.
As you can see, lame adds some severe aliasing distortion. At it's
worst, it's at about -24dB. The original sine wave was at -6dB. The
distortion is at 14.5kHz which means it aliased from 17.5kHz -
therefore whatever anti-alias filter lame is using has a -18dB
rejection ratio at 17.5kHz. It should be down to -96dB or lower by then.
What does this sound like? Well, on a simple test signal like this,
the most audible problem is that as the frequency of the test tone
increases beyond 16kHz, rather than continuing to increase beyond the
range of your hearing, it starts coming down again. The ultrasonic
part you shoudln't be able to hear is reflected back into the region
where you CAN hear it. This is bad.
The other problem is that you're effectively adding spectral
components to the signal. This makes the job of an mp3 encoder harder
- which ones should it discard?
I used lame 3.86BETA for this test. CEP took 1 minute to resample this
20 second test file. By using a lower quality setting (256), it can do
it in real time, and still do OK. Lowering the quality (increasing the
speed) further degrades its performance. Lame can resample AND encode
the signal in 9 seconds, which explains why it does such a poor job.
Maybe someone working on lame can add an optional high quality
resampling routine? Can someone on the mail list suggest it?
Hope this is some interest. If it's not, ignore me! Any questions are
welcome.
David. http://www.David.Robinson.org/mp3decoders/
The truth about mp3 decoding quality updated.
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