Hi, I know the subject line makes it prone to immediate dismissal with the
(absolutely true) comment "bad idea", but let me expalin.
I've been reading this mailing list for a while in the past few months but
since my DSP experience is fading away and I have never really gotten into
studying the mp3 format down to the technical details I've just been lurking
from the outside.
My problem arises from the desire to record streaming audio, such as
RealAudio or WMA formats. Apart some rare exceptions the only way
to do it is by recording the sound card output with programs such as Total
Recorder (www.highcriteria.com -BTW the latest version allows on the fly mp3
encoding using LAME_ENC.dll, it should probably be added to the LAME
homepage-). The stream is recorded before going through the sound card so
there are no artifacts from A/D->D/A conversions and the output is a regular
wav file.
Of course recompression is a bad idea, but in this case it is a necessity.
The
goal would be to preserve the quality of the decompressed file
while keeping it at a reasonable size. Following the suggestions on
www.r3mix.net I've used the VBR settings recommended there, but I've come to
think that they're overkill for this kind of application.
>From the only technical reference I could find on RealAudio, the specs for
their hi-fi codecs are as follow:
44 Kbps Stereo Music: Sampling Freq. 22.05 kHz , freq. response 11 kHz
64 Kbps Stereo Music: Sampling Freq. 44.1 kHz , freq. response 16 kHz
96 Kbps Stereo Music: Sampling Freq. 44.1 kHz, freq. response 20 kHz
So we know the bitrate of the original file but of course their psycho model
and proprietary tricks are unknown, therefore I wonder what assumptions can
be made to tweak the LAME settings. For example is it reasonable to assume
they can't be more than 50% more efficient than mp3 and set the minimum
bitrate to ~1.5 x their original bitrate? or use ABR with their bitrate (or
1.5x) as the target? LAME has a very weird behavior with these files, most
of the times it either stays at the minumum bitrate (that alone is good
evidence that the quality is set too high) or (especially when encoding
files sampled at 22KHz) it stays almost constantly at 160Kbps.
Any ideas? Does anybody have any more technical details about the algorithms
used by realaudio or WMA?
thanks in advance, any suggestion will be greatly appreciated.
GD
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