On Wednesday, March 14, 2012 7:44:03 AM UTC-4, Spooky wrote:
>
> You didn't mention your sampling frequency....  Remember Nyquist?  To
> reproduce any digital signal, you MUST sample at twice the frequency of
> the highest frequency you want to reproduce.  In this case, you'd have to
> sample at a minimum of 34,000 samples/second.  HOWEVER.....
>
Not exactly true.  What is true is that frequency components above the 
threshold appear like, and cannot be distinguished from, frequencies below 
it.  To avoid this confusion, it is common to remove them with a low pass 
filter before sampling.  However, this is not the only way of doing things. 
 It is also quite common to use a filter with selects frequencies above the 
threshold, and take advantage of the sampling folding them down to a 
frequency range where they are easier to deal with computationally. 
 Obviously that's more common in communications equipment where you are 
only interested in a narrow range of frequencies, rather than baseband 
voice or music.
 

> > When I analyze the recorded sound, 17kHz frequencies aren't there, its
> > like the phone has a low pass filter that eliminates these
> > frequencies.
>
> Yes, the *PHONE* certainly does...but where?  The question is, does the
> non-phone portion of the Android device have access to the raw audio, or
> has the low-pass filter already been applied?  Check to see if you can
> record frequencies above 5 kHz (DS0[1] is 300--3400 Hz, but since there
> are no "brick-wall" filters, it usually extends to about 4000 Hz, and
> then we allow a bit extra to make sure).  If you can access up to 5000 Hz
> or higher, you're getting audio before the low-pass filter.  
>

Before *which* low pass filter?   A device intended to be used for both 
telephony and higher quality audio likely has several.  There is probably 
something in front of the actual ADC (likely on chip) which prevents 
aliasing in the sampling itself.  There may be an intermediate filter in 
the audio system if any resampling is done, and this may depend on the 
sample rate requested by the program.  And then there's probably one in the 
GSM or SIP or whatever encoding system to limit to the narrower voice 
bandwidth of those channels.

 

>

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