5 Apr
2006
5 Apr
'06
9:27 p.m.
Tobias Scharnberg wrote:
Hello List,
I'm trying to find a library or code-snippet in order to do audio
resampling from 8khz to 44,1khz and from 44,1khz to 8khz. I need to
resample the data in realtime - resampling a buffer of data, not a
soundfile. The quality doesn't need to be good so I guess the best
solution might be linear audio resampling. The device to do the
resampling on is an ARM CM-X255 running at 400MHz.
I tried out libsamplerate so far but when I tested it with the
soundfile conversion test program it needed 3,5 secs to sample from
8kHz to 44,1 khz for a 1,7 secs audiofile - which is too slow for me.
Is there something faster that can do the job?
Any suggestions are highly welcome
Hi Tobias
I suggest using a Kaiser-window based interpolation algorithm. This is
the best filter method to use for sample rate conversion. You can get
decent results with 8 taps or even fewer. For example you can get a 6dB
bandwidth of about 2.8kHz (at Fs=8kHz) with 50dB alias rejection. (of
course you need more taps for high-quality results)
I implemented an 8 tap SRC on an ARM-9 using integer arithmetic... I
can't remember the exact CPU usage but it was only of the order of 10MHz
or so. I'm not familiar with the CPU you mention - if it hasn't got the
DSP instructions then it will be a bit slower but you seem to have
plenty to spare!
Linear interpolation is no good for audio - you will have about 10%
distortion!
Andrew