I heard somewhere that Fourier transform only works on a signal that
contains parts that are harmonic to each other (i.e. f, 2f, 3f etc.).
No. What you say would imply that the signal must be periodic, and that is not true. The Fourier transform can be computed for any signal, periodic or not, with finite energy or even with finite power. In the latter case (of finite power), the transform will, in most cases, be unbounded.
What you say is correct for Fourier series, not for the Fourier transform. The Fourier series decomposition is defined only for periodic signals. The spectrum of a periodic signal will be able to have contents only at \$f_0\$, \$2f_0\$, \$3f_0\$, etc. That does not mean that it will always have contents at all those frequencies. It means that it cannot have contents outside that set of frequencies.
Can anyone explain me on proof of this?
That was not right, so it cannot be proved.
What is the sufficient condition for Fourier transform?
There is no "the sufficient condition". There are several sufficient conditions.
For instance, one sufficient condition is that it satisfies both Dirichlet conditions:
1) Over any time interval of finite length, the function w(t) is single valued with a finite number of maxima and minima, and the number of discontinuities (if any) is finite.
2) w(t) is absolutely integrable.
That condition is sufficient, but not necessary.
A weaker sufficient condition for the existence of the Fourier transform is that the signal has finite energy. All physically realizable waveforms are finite-energy, so that means that all physical waveforms encountered in engineering practice are Fourier transformable. However, that is to have a Fourier transform with bounded values. If you allow yourself to work with unbounded values, then you can also compute the Fourier transform of a signal that has finite power (which is a less strict requirement). For instance, a sin(\$2\pi f_0·t\$) that exists for all t has finite power, but not finite energy. You can compute the Fourier transform of such a signal, but it will have unbounded values (in fact, "deltas") at \$f_0\$ and \$-f_0\$.
edited for more depth & clarity to avoid artifacts, pun intended
You can control your artifacts with an improved "anti-alias" filter to eliminate all content of signals above the Nyquist rate. This is often more critical if the image has been digitized already by another system at some unknown pixel rate.
For example a high resolution scan of magazine photo without the appropriate anti-alias filter setting will result in herringbone artifacts that are much worse than the original. This is a similar effect to the stripes of a shirt on TV with harmonic pixel interference herringbone patterns or the fringing patterns of lenses on a digital camera or camcorder in the presence of infrared light and similar resolution subject matter above the pixel resolution of the camera.
How you implement the anti-alias filter depends on the original signal.
- In the more general sense, if the original signal is natural light to be captured for 2D imaging, you might use an AA optical "blur filter".
- If the original signal is light reflected from a digital printed object in a scanner, you might tune the alias digital blur filter to match the resolution of the original print and synchronously if possible. e.g. 100 dpi for fax content
- If the original signal is a 3D scan using magneto-resonant imaging then you need to determine is it from stop-band remnant signals or pass-band group delay distortion.
The quality factors of low artifacts depend on the relative levels of interference from each to avoid alias artifacts.
Over-sampling makes the job of ideal bandstop filtering easier but ends up with huge file sizes so this can be followed by undersampling to achieve easier brick wall filters with desired blur filter in pass band.
Wiki has a more examples of various anti-alias filters and also spacial filtering
applications.
Best Answer
Yes. If the signal contains higher frequency parts, aliasing will occur and the signal will be distorted, this is why all digital recording equipment has filters that remove the signal parts above f/2. It is also why audio is recorded using at least 44kHz and not 40. If you recorded sound using 40kHz sampling rate and still wanted the frequency response to xtend to 20kHz you would need a filter with infinitely steep slope (pass 19.999kHz, completely block 20.000kHz).
It is also why recording using higher sampling rates is better - it allows the device to have a filter with less steep slope (for example, pass 20kHz, completely block 48kHz in case of 96kHz sampling rate) so there are less phase distortions.
If the signal has limited bandwidth and does not start at 0Hz (radio signals for example), then the sampling rate has to be greater than twice the bandwidth, but not maximum frequency (if the minimum frequency component is 1MHz and maximum is 1.1MHz then sampling rate has to be higher than 200kHz). For audio, the maximum frequency is the same as bandwidth, since audio usually starts at very close to 0Hz.