I have a need to convert audio samples from 11025 and 22050 to 44100; I'm looking for the fastest and best sounding conversion routine. I require that the answer be given in pure Java, without the need for external routines or libraries. The source is an array of short values representing left and right channels, interleaved like so LRLRLRLR
I've heard that gaussian transformation is the best, but it is a cpu killer.
Update
To add more detail, I would like a mix between best and fastest. The answer would give great sounding audio suitable for near real-time communication.
Update 2
I'm looking for some short code examples for this one, should be ez points for you audio guru's
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132answers:
2
+1
A:
Well, it is hard to resample it slow enough so that it is not realtime :-) One of the best & still fast solutions is to do forward FFT, and then do reverse FFT with any sample rate you need.
You may implement this on your own, or copy & paste any FFT implementation.
This might work like 100x realtime or faster, not sure you need 1000x faster (in this case you can go for linear or bicubic interpolation) :-)
BarsMonster
2010-10-11 08:57:22
A:
you can (ultimately) just use a fir after filling every other sample with 0s - you're upsampling by 2 or 4. this will be plenty fast for realtime. audio quality will be fine for most applications.
Justin
2010-10-11 09:21:12
I'm not sure, but filling every other sample with a zero would probably sound *very odd*... It should probably duplicate the values.
Frank
2010-10-16 03:26:34
Even after applying a FIR (http://en.wikipedia.org/wiki/Finite_impulse_response) filter to the embedded zeroes, wouldn't that still sound odd?
Roland Illig
2010-10-17 00:22:10
@Frank and Roland Illing you may not realize that duplicating the samples (assuming a signal upsampled by an integer which has been zero-filled) could be achieved perfectly with a very simple fir: all values=1, length=upsampling amount (2 or 4 in this case). in practice, it's better to use a longer window function (e.g., sinc or Hamming). the filter fills the gaps, and serves as an interpolator. which function is 'best' depends on the signal, and the characteristics you want (or don't).
Justin
2010-10-19 06:47:10