## Fourier Theorems for the DTFT

This section states and proves selected Fourier theorems for the DTFT. A more complete list for the DFT case is given in [264].^{3.4}Since this material was originally part of an appendix, it is relatively dry reading. Feel free to skip to the next chapter and refer back as desired when a theorem is invoked.

As introduced in §2.1 above, the Discrete-Time Fourier Transform (DTFT) may be defined as

(3.8) |

We say that is the

*spectrum*of .

### Linearity of the DTFT

(3.9) |

or

(3.10) |

where are any

*scalars*(real or complex numbers), and are any two

*discrete-time signals*(real- or complex-valued functions of the integers), and are their corresponding continuous-frequency

*spectra*defined over the unit circle in the complex plane.

*Proof:*We have

### Time Reversal

For any complex signal , , we have(3.11) |

where .

*Proof:*

(3.12) |

denote such a definition. Then in this case we have

(3.13) |

*Proof:*

(3.14) |

In the typical special case of real signals ( ), we have so that

(3.15) |

That is,

*time-reversing a real signal conjugates its spectrum*.

### Symmetry of the DTFT for Real Signals

Most (if not all) of the signals we deal with in practice are real signals. Here we note some spectral symmetries associated with real signals.#### DTFT of Real Signals

The previous section established that the spectrum of every real signal satisfies(3.16) |

*I.e.*,

(3.17) |

In other terms, if a signal is real, then its spectrum is

*Hermitian*(``conjugate symmetric''). Hermitian spectra have the following equivalent characterizations:

- The real part is even, while the imaginary part is odd:

- The magnitude is even, while the phase is odd:

#### Real Even (or Odd) Signals

If a signal is*even*in addition to being real, then its DTFT is also real and even. This follows immediately from the Hermitian symmetry of real signals, and the fact that the DTFT of any even signal is real:

*I.e.*,

*odd*and real, then its DTFT is odd and

*purely imaginary*. This follows from Hermitian symmetry for real signals, and the fact that the DTFT of any odd signal is imaginary.

### Shift Theorem for the DTFT

We define the*shift operator*for sampled signals by

(3.18) |

where is any integer ( ). Thus, is a

*right-shift*or

*delay*by samples. The

*shift theorem*states

^{3.5}

(3.19) |

or, in operator notation,

(3.20) |

*Proof:*

*linear phase term*, so called because it is a linear function of frequency with slope equal to :

(3.21) |

The shift theorem gives us that multiplying a spectrum by a linear phase term corresponds to a

*delay*in the time domain by samples. If , it is called a time

*advance*by samples.

### Convolution Theorem for the DTFT

The*convolution*of discrete-time signals and is defined as

(3.22) |

This is sometimes called

*acyclic convolution*to distinguish it from the

*cyclic convolution*used for length sequences in the context of the DFT [264]. Convolution is cyclic in the time domain for the DFT and FS cases (

*i.e.*, whenever the time domain has a finite length), and acyclic for the DTFT and FT cases.

^{3.6}The

*convolution theorem*is then

(3.23) |

That is,

*convolution in the time domain corresponds to pointwise multiplication in the frequency domain*.

*Proof:*The result follows immediately from interchanging the order of summations associated with the convolution and DTFT:

### Correlation Theorem for the DTFT

We define the*correlation*of discrete-time signals and by

*correlation theorem*for DTFTs is then

*Proof:*

### Autocorrelation

The*autocorrelation*of a signal is simply the cross-correlation of with itself:

(3.24) |

From the

*correlation theorem*, we have

*sample autocorrelation*to include a normalization suitable for this case (see Chapter 6 and Appendix C). From the autocorrelation theorem we have that a digital-filter impulse-response is that of a

*lossless allpass filter*[263] if and only if . In other words, the autocorrelation of the impulse-response of every allpass filter is impulsive.

### Power Theorem for the DTFT

The*inner product*of two

*signals*may be defined in the time domain by [264]

(3.25) |

The inner product of two

*spectra*may be defined as

(3.26) |

Note that the frequency-domain inner product includes a normalization factor while the time-domain definition does not. Using inner-product notation, the

*power theorem*(or

*Parseval's theorem*[202]) for DTFTs can be stated as follows:

(3.27) |

That is, the inner product of two signals in the time domain equals the inner product of their respective spectra (a complex scalar in general). When we consider the inner product of a signal with itself, we have the special case known as the

*energy theorem*(or

*Rayleigh's energy theorem*):

(3.28) |

where denotes the norm induced by the inner product. It is always real.

*Proof:*

### Stretch Operator

We define the*stretch operator*in the

*time domain*by

(3.29) |

In other terms, we stretch a sampled signal by the factor by inserting

*zeros*in between each pair of samples of the signal. In the literature on multirate filter banks (see Chapter 11), the stretch operator is typically called instead the

*upsampling*operator. That is, stretching a signal by the factor of is called upsampling the signal by the factor . (See §11.1.1 for the graphical symbol ( ) and associated discussion.) The term ``stretch'' is preferred in this book because ``upsampling'' is easily confused with ``increasing the sampling rate''; resampling a signal to a higher sampling rate is conceptually implemented by a stretch operation followed by an ideal lowpass filter which moves the inserted zeros to their properly interpolated values. Note that we could also call the stretch operator the

*scaling*operator, to unify the terminology in the discrete-time case with that of the continuous-time case (§2.4.1 below).

### Repeat (Scaling) Operator

We define the*repeat operator*in the

*frequency domain*as a

*scaling*of frequency axis by some integer factor :

(3.30) |

where denotes the radian frequency variable after applying the repeat operator. The repeat operator maps the entire unit circle (taken as to ) to a segment of itself , centered about , and repeated times. This is illustrated in Fig.2.2 for .

Since the frequency axis is continuous and -periodic for DTFTs, the repeat operator is precisely equivalent to a scaling operator for the Fourier transform case (§B.4). We call it ``repeat'' rather than ``scale'' because we are restricting the scale factor to positive integers, and because the name ``repeat'' describes more vividly what happens to a periodic spectrum that is compressively frequency-scaled over the unit circle by an integer factor.

### Stretch/Repeat (Scaling) Theorem

Using these definitions, we can compactly state the*stretch theorem*:

(3.31) |

*Proof:*

*ideal sampling-rate conversion*for integer upsampling ratios : We first stretch the signal by the factor (introducing zeros between each pair of samples), followed by an

*ideal lowpass filter*cutting off at . That is, the filter has a gain of 1 for , and a gain of 0 for . Such a system (if it were realizable) implements

*ideal bandlimited interpolation*of the original signal by the factor . The stretch theorem is analogous to the

*scaling theorem*for continuous Fourier transforms (introduced in §2.4.1 below).

### Downsampling and Aliasing

The*downsampling*operator selects every sample of a signal:

(3.32) |

The

*aliasing theorem*states that downsampling in time corresponds to

*aliasing*in the frequency domain:

(3.33) |

where the operator is defined as

(3.34) |

for . The summation terms for are called

*aliasing components*. In

*z*transform notation, the operator can be expressed as [287]

(3.35) |

where is a common notation for the primitive th root of unity. On the unit circle of the plane, this becomes

(3.36) |

The frequency scaling corresponds to having a sampling interval of after downsampling, which corresponds to the interval prior to downsampling. The aliasing theorem makes it clear that, in order to downsample by factor without aliasing, we must first lowpass-filter the spectrum to . This filtering (when ideal) zeroes out the spectral regions which alias upon downsampling. Note that any rational sampling-rate conversion factor may be implemented as an upsampling by the factor followed by downsampling by the factor [50,287]. Conceptually, a stretch-by- is followed by a lowpass filter cutting off at , followed by downsample-by- ,

*i.e.*,

(3.37) |

In practice, there are more efficient algorithms for sampling-rate conversion [270,135,78] based on a more direct approach to

*bandlimited interpolation*.

#### Proof of Aliasing Theorem

To show:(3.38) |

where we have chosen to keep frequency samples in terms of the original frequency axis prior to downsampling,

*i.e.*, for both and . This choice allows us to easily take the limit as by simply replacing by :

(3.39) |

Replacing by and converting to -transform notation instead of Fourier transform notation , with , yields the final result.

### Differentiation Theorem Dual

**Theorem:**Let denote a signal with DTFT , and let

(3.40) |

denote the derivative of with respect to . Then we have

*Proof:*Using integration by parts, we obtain

**Corollary:**Perhaps a cleaner statement is as follows:

**Next Section:**

Continuous-Time Fourier Theorems

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Fourier Transform (FT) and Inverse