Hi,
I need to upsample the audio data @ 24KHz to 48KHz.
One simple method I know is to do "zero stuffing" between the two samples and then Low Pass Filter it with cut-off of the filter at pie/2.

This method on implementation(in C and then to put it on a DSP) gives a large complexity.
Can you suggest me some method with the least complexity.
Is there a single filter(FIR or else) design to do both above mentioned tasks??
Or, if you could tell me how is upsampling(like one I need to do) is implemented in professional audio codecs.

Please reply it ASAP.

Thanks

Regards,
Devilal

moved.

I think real world resamplers zero pad/decimate then filter.

I don't know much about this, but I'd guess if FIR is too slow you could try IIR. They're generally quite fast but harder to work with and can do ugly things with phase.

Hopefully someone more experienced can fill in the details.

from General Audio to General Audio?
http://www.hydrogenaudio.org/forums/index....showtopic=46157

Btw. this is a violation of TOS #6 (double post).

admin, close or remove this thread, the real discussion was started in the other tread.

sebastian; from general a/v to general audio:
http://www.hydrogenaudio.org/forums/index.php?showforum=53

You could replace the added zeros with an if clause wich pretends that there are zeros and then run the filter. This saves ram.

And the method with zeros and FIR or IIR filter is the only one I know.

Or you could take the filtering to frequence domain via Fourier transform.