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Hi all,
I am trying to resample a 22.05 khz audio signal(.wav
file) to 12.8 khz.In doing that the interpolation
and decimation factors will bevery high.
Is there any other approach to adjust decimation and
interpolation factors to shorter values.
Suppose i have 20ms of wav file samples sampled at
22.05 khz how can i change the sampling frequebcy to
12.8 khz.
Please don't suggest solutions in MATLAB  as i want to
implement the above thing in C.
Thanks and Regards,
Mayank
	

(You need to be a member of audiodsp -- send a blank email to audiodsp-subscribe@yahoogroups.com )

A simple solution is the following:
1. Design an oversampling FIR low pass filter: e.g. Fs = 16 * 22.05kHz = 352.8kHz, Pass band = 5.0 to 6.0kHz as required and Stop band = 12.8kHz / 2 = 6.4kHz, 70 or 80dB attenuation as required.
By using this as a poly-phase structure, for each sample at 22.05kHz, 16 possible phases are available.
2. Use an interpolator to obtain your 12.8kHz output samples from 2 adjacent phases (effectively only 1/8 of the filter taps are used per output sample).

Sorry for the short answer, time is short.

Good luck,

Alex
DSP software Engineer
Consultant for Philips Applied Technologies


Hi all,
I am trying to resample a 22.05 khz audio signal(.wav
file) to 12.8 khz.In doing that the interpolation
and decimation factors will bevery high.
Is there any other approach to adjust decimation and
interpolation factors to shorter values.
Suppose i have 20ms of wav file samples sampled at
22.05 khz how can i change the sampling frequebcy to
12.8 khz.
Please don't suggest solutions in MATLAB  as i want to
implement the above thing in C.
Thanks and Regards,
Mayank



(You need to be a member of audiodsp -- send a blank email to audiodsp-subscribe@yahoogroups.com )