Hello,
Is is possible to do automatic upsampling of MP3's? I mean, if for instance a MPEG file is 32kHz, the output can be 44,1kHz anyway. Unless I missed it, I had the impression it was not possible and have to go looking for other sources.
Regards,
Armin
On Tuesday, May 28, 2002, at 05:17 AM, armin.gerritsen@philips.com wrote:
There's been some interest in whether something like this can be done as part of the decoding process, during synthesis, but I have yet to fully research the possibility. For the moment the only alternative is to explicitly resample after synthesis.
-- Rob Leslie rob@mars.org
Rob Leslie wrote:
There are several Linux audio drivers that do this when faced with hardware that will only operate at one (or a few) sampling frequencies. They simply replicate (or mabye try to linear interpolate) samples such that over a small time window the start and end samples are properly timed. Since it's the final output stage, it seems to be a sufficient solution.
-- Dan
While linear interpolation or replicating samples is okay for small computer speakers or cheap headphones, even non-audiophiles like me can tell the difference between poor resampling and good resampling, when I've good good speakers to reproduce the sound.
There are a lot of free resampling algorithms out there, but none of them are drop-in libraries that are easy to use in an existing program (like libmad is). A Sourceforge project was recently started to address that problem. I think that their API is actually very nice - and it would make it possible for tools like libmad to resample "on the fly". So far the source code they've made available just has a single implementation based on linear interpolation, but more methods are coming.
http://libsample.sourceforge.net/
- Dominic
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armin.gerritsen＠philips.com -
Dan Malek -
Dominic Mazzoni -
Rob Leslie