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frequency band shift...

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pooja_gajjar

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hey everyone !!!

i need some help in matlab coding of my project....
actually i have wideband speech signal of frequency range from 0-8 khz...
i have devided my signal in low(0-3.4) and high band(3.4-8 khz)
and the next is to shift that high band signal (3.4-8 khz) to lower band range (0-3.4 khz)
i need help that how can i shift that higher band to lower in matlab ????
please if anyone have idea then suggest me ....
hoping for help...
 

Not clear the purpose of that.

However, if I would need do that, I first would convert input data from Time domain to Frequency domain ( i.e: Fourier Transform ).
Applying a linear conversion, could now shift bandwidth : F(f) = ( ( f -3.4 ) / ( (8-3.4)/3.4 ) )

Thus, transforming now input data from Frequency domain to Time domain, we can retrieve data translated to desired frequency.


+++
 

Andre pretty much covered your query, am just adding some matlab stuff you could use..

take the input x signal in time domain
where L satisfies your sampling requirement.
y=fft(x,L)

%Now in Frequency domain:

In regards to translation..it seems your compressing your higher frequency band into a smaller bandwidth.
If this is the case you can probably follow "andre" ratioed equation.

If you typically have higher frequency noise form your signal you could simply shift linearly by 3.4, i.e. and avoid the distortion.

Have Fun
 

hey thank you for help.... and i got your point but how should i define value of "f" ???
also matlab will accept integer index value of frequency as i know !!!!
please if have any idea....
 

Start thinking in the type of procesing you need. Once your concept is clear you can go to implementation (Matlab or any other).
The goal is to transmit medium fidelity audio using two voiceband channels?
It is a system that should be able to run continuously in real time, and not just a test/simulation with a short amount if data, right?

You can translate the upper band to lower using SSB (single side-band modulation).
The problem is that in SSB the bandwidth is conserved: you can not compress bandwidth without using some sophisticated coding scheme. But you can accomodate into the 300Hz-3400Hz band the 3400-6500 Hz band. It's no bad.

Regards

Z
 
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pooja_gajjar said:
...how should i define value of "f" ???...
Click to expand...

The first step, is to sample a set of values in time domain : X=(x1,x2,x3...xn).
So, apply this at FFT funcion as above sugested by kalyanasv.
In frequency domain, each data of vector F=(f1,f2,f3....fn) will be operated by the previous postulated formula.

f7900.gif


The DSP process is done continuouslly : A/D(t) --> FFT(t,f) --> SHIFT/SCALE(f) --> INV_FFT(f,t).
This filtering is performed step-by-step, taking an entyre set of data of the vector F sampled.

+++
 

I think the SSB method is the only one that could be applied without additional specifications (if we accepct to cut a part of the frequency range). For all others, that can't work continously by nature, e.g. FFT based, you need to define block lengths, windows, overlapping factors etc.
 

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