# Discrete Time Fourier Transform and Discrete Fourier

### Contents

**Describe DTFT and its properties**1**Describe DTFT of Common Signals**2**Describe Parseval’s Theorem**3**Explain Inverse DTFT**3**Explain Convergence of DTFT**4**Describe Discrete Time Hilbert Transform**5**Define DFT**6**Describe properties of DFT and its Applications**7**Describe Convolution and its effect in the frequency domain**8**Describe Discrete Fourier Series (DFS)**9**Explain Multiplication of 2DFT’s**9**Describe Circular Convolution**10

Describe DTFT and its properties

DTFT stands for Discrete-Time Fourier Transform, and it is a mathematical tool used to analyse and represent the frequency components of a discrete-time signal. The DTFT of a discrete-time signal x[n] is given by:

X(e^(jω)) = Σn= -∞ to ∞ x[n] e^(-jωn)

where ω is the normalised frequency and ranges from -π to π.

The properties of DTFT are:

- Linearity: DTFT is linear, which means if a signal x[n] is a linear combination of other signals, then its DTFT is also a linear combination of their DTFTs.
- Time shifting: If x[n] is shifted by k samples, then its DTFT is multiplied by e^(-jωk).
- Frequency shifting: If x[n] is multiplied by e^(jω0n), then its DTFT is shifted by ω0.
- Time reversal: If x[n] is reversed, then its DTFT is conjugated, i.e., X(e^(jω)) = X*(e^(-jω)).
- Time scaling: If x[n] is scaled by a factor M, then its DTFT is compressed by a factor of 1/M.
- Convolution: The DTFT of a convolution of two signals is equal to the product of their DTFTs.
- Periodicity: If x[n] is periodic with period N, then its DTFT is also periodic with period 2π/N.

Describe DTFT of Common Signals

The Discrete-Time Fourier Transform (DTFT) is a mathematical tool used to analyse the frequency content of discrete-time signals, which are signals that are sampled at a fixed rate. The DTFT can be used to determine the frequency-domain representation of common signals, as follows:

- Unit impulse signal: The DTFT of a unit impulse signal is a constant value of 1 for all frequencies.
- Unit step signal: The DTFT of a unit step signal is a complex function that has a magnitude of zero for frequencies less than zero, and a magnitude of infinity for frequencies greater than zero. The phase of the DTFT is zero for frequencies less than zero, and pi for frequencies greater than zero.
- Sinusoidal signal: The DTFT of a sinusoidal signal with frequency w0 is a pair of impulses located at frequencies w0 and -w0, with magnitudes that depend on the amplitude and phase of the sinusoidal signal.
- Complex exponential signal: The DTFT of a complex exponential signal with frequency w0 is a pair of impulses located at frequencies w0 and -w0, with magnitudes and phases that depend on the amplitude and phase of the complex exponential signal.
- Rectangular pulse signal: The DTFT of a rectangular pulse signal is a sinc function, which is a sinusoidal function that decays to zero at higher frequencies.
- Triangular pulse signal: The DTFT of a triangular pulse signal is a sinc squared function, which is a sinusoidal function that decays to zero at higher frequencies more slowly than the sinc function.

These are some of the common signals and their corresponding DTFTs. The DTFT can be used to analyse the frequency content of any discrete-time signal, and is a useful tool for signal processing and analysis.

Describe Parseval’s Theorem

Parseval’s theorem is a mathematical theorem that relates the energy or power of a signal in the time domain to its frequency domain representation. Specifically, Parseval’s theorem states that the energy or power of a signal can be computed as the sum of the squares of the magnitudes of its Fourier transform coefficients.

In mathematical notation, if x(n) is a discrete-time signal and X(e^(jw)) is its Fourier transform, then Parseval’s theorem can be expressed as:

- For energy: Sum of squares of samples in time domain = (1/2*pi) * Integral of squared magnitude of X(e^(jw)) over all frequencies from -pi to pi
- For power: Average power of signal in time domain = (1/2*pi) * Integral of squared magnitude of X(e^(jw)) over all frequencies from -pi to pi

Here, the square of the magnitude of X(e^(jw)) is also known as the spectral density or power spectral density of the signal, and represents the distribution of power over different frequencies.

Parseval’s theorem is important because it provides a way to compute the energy or power of a signal using its frequency-domain representation, which can be more convenient or efficient in some cases. It is also useful for verifying that a signal processing operation has not changed the overall energy or power of a signal.

Explain Inverse DTFT

The Inverse Discrete-Time Fourier Transform (IDTFT) is a mathematical tool used to recover a discrete-time signal from its frequency-domain representation, which is given by the Discrete-Time Fourier Transform (DTFT).

In mathematical notation, if X(e^(jw)) is the DTFT of a discrete-time signal x(n), then the DTFT is given by:

- x(n) = (1/2*pi) * Integral of X(e^(jw)) * e^(jwn) over all frequencies from -pi to pi

In other words, the DTFT involves taking the inverse Fourier transform of the DTFT by integrating over all frequencies from -pi to pi. The resulting function x(n) is the original discrete-time signal, and represents its time-domain representation.

The DTFT is an important tool in signal processing because it allows us to analyse signals in both the time and frequency domains. Specifically, we can use the DTFT to analyse the frequency content of a signal, and the DTFT to recover the original signal from its frequency-domain representation. This is useful for applications such as filtering, where we may want to modify a signal in the frequency domain and then recover the modified signal in the time domain.

It is worth noting that not all frequency-domain representations have an inverse DTFT. In order for the DTFT to exist, the DTFT must satisfy certain conditions, such as being absolutely summable or having a bounded magnitude.

Explain Convergence of DTFT

The Discrete-Time Fourier Transform (DTFT) is a mathematical tool used to analyse the frequency content of discrete-time signals. However, the DTFT may not converge for all signals, which can limit its usefulness in some cases.

The convergence of the DTFT depends on the properties of the discrete-time signal being analysed. Specifically, the DTFT will converge if the signal is absolutely summable, which means that the sum of the magnitudes of all samples is finite.

In mathematical notation, if x(n) is a discrete-time signal and X(e^(jw)) is its DTFT, then the DTFT converges if:

- Sum of |x(n)| from n = -infinity to infinity is finite

When the DTFT converges, it is also continuous and periodic with a period of 2*pi. The frequency-domain representation X(e^(jw)) is a complex-valued function that can be plotted as a continuous curve over the range of frequencies from -pi to pi.

If the DTFT does not converge, it is not well-defined and cannot be used to analyse the frequency content of the signal. In some cases, it may be possible to modify the signal in order to make the DTFT converge. For example, we can apply a windowing function to the signal to reduce its amplitude at the edges, which can improve convergence.

Overall, the convergence of the DTFT is an important consideration when using this tool for signal processing and analysis. Signals that are absolutely summable can be analysed using the DTFT, while signals that are not absolutely summable may require alternative techniques for frequency-domain analysis.

Describe Discrete Time Hilbert Transform

The Discrete Time Hilbert Transform (DTH) is a mathematical operation that is used to compute the analytic representation of a discrete-time signal. The analytic representation of a signal consists of two parts: the original signal and its Hilbert transform, which is a version of the signal that is shifted in phase by 90 degrees.

In mathematical notation, if x(n) is a discrete-time signal, then its analytic representation is given by:

- x_a(n) = x(n) + j*h[n]

Where x_a(n) is the analytic representation of x(n), j is the imaginary unit, and h[n] is the discrete-time Hilbert transform of x(n).

The discrete-time Hilbert transform can be computed using the following formula:

- h[n] = (1/pi) * P.V. Integral of x(k)/(n-k) dk

Where P.V. indicates the principal value of the integral. This formula expresses the Hilbert transform as a convolution of the signal x(n) with the sequence 1/(pi*n), which is known as the Hilbert kernel.

The DTH has many important applications in signal processing, particularly in the analysis of communication signals. By computing the analytic representation of a signal, we can separate the positive and negative frequency components of the signal, which can be useful for frequency-modulation and demodulation applications. The DTH can also be used to compute the instantaneous amplitude and phase of a signal, which are important for applications such as signal demodulation and signal filtering.

Overall, the DTH is a powerful tool for signal processing that allows us to compute the analytic representation of a discrete-time signal and separate its positive and negative frequency components.

Define DFT

DFT stands for “Discrete Fourier Transform,” which is a mathematical technique used to convert a time-domain signal, such as a digital audio signal, into its frequency-domain representation. The DFT is a widely used tool in digital signal processing, communication systems, and other fields.

The DFT takes a finite sequence of complex numbers (usually representing a discrete-time signal) and returns another finite sequence of complex numbers of the same length, which represents the signal’s spectral content. The spectral content indicates the amplitude and phase of each frequency component present in the original signal.

The DFT is an important tool because it allows us to analyse and manipulate signals in the frequency domain, where many signal processing techniques, such as filtering and equalisation, are more easily implemented. It is also used in many applications, such as audio and image compression, data compression, and wireless communication systems.

Describe properties of DFT and its Applications

Properties of DFT:

- Linearity: The DFT is a linear operation. That is, if we apply the DFT to a linear combination of two signals, we get the same linear combination of their individual DFTs.
- Time shifting: The DFT of a time-shifted signal is obtained by multiplying the DFT of the original signal by a complex exponential function.
- Convolution: The DFT of the convolution of two signals is equal to the product of their individual DFTs.
- Periodicity: The DFT of a periodic signal is a set of discrete spectral lines that repeat at integer multiples of the fundamental frequency.

Applications of DFT:

- Spectral analysis: The DFT is used to analyse the frequency content of a signal and identify its spectral components.
- Filtering: The DFT is used in digital signal processing to implement filters that can remove or attenuate certain frequency components of a signal.
- Compression: The DFT is used in audio and image compression algorithms to identify and remove redundant or irrelevant spectral components, leading to a smaller data size.
- Channel estimation: The DFT is used in wireless communication systems to estimate the frequency response of a communication channel and compensate for its distortions.
- System identification: The DFT is used to identify the impulse response of a system from its input and output signals.
- Frequency-domain equalisation: The DFT is used to implement equalisation techniques in communication systems, such as adaptive equalisation and linear equalisation.

Describe Convolution and its effect in the frequency domain

Convolution is a mathematical operation that is used to describe the interaction between two signals or systems. In the context of digital signal processing, convolution is a way to compute the output of a linear time-invariant (LTI) system when it is given an input signal.

The convolution of two signals x(n) and h(n) is defined as the sum of the products of each sample of x(n) with a time-reversed and shifted version of h(n). That is:

y(n) = sum over k from -inf to inf (x(k) * h(n-k))

In the frequency domain, the convolution of two signals corresponds to the product of their individual Fourier transforms. That is:

Y(f) = X(f) * H(f)

Where Y(f), X(f), and H(f) are the Fourier transforms of y(n), x(n), and h(n), respectively, and * denotes complex multiplication.

The effect of convolution in the frequency domain can be understood by considering the frequency response of an LTI system. The frequency response of an LTI system is the Fourier transform of its impulse response, which is the output of the system when it is given a unit impulse input.

When we convolve a signal x(n) with an impulse response h(n), we are effectively filtering x(n) with the frequency response of the system described by h(n). In the frequency domain, this corresponds to multiplying the Fourier transform of x(n) by the Fourier transform of h(n), which is the frequency response of the system.

Therefore, convolution in the time domain corresponds to multiplication in the frequency domain, and vice versa. This property is known as the convolution theorem and is a powerful tool for analysing linear systems and designing filters.

Describe Discrete Fourier Series (DFS)

Discrete Fourier Series (DFS) is a mathematical technique used to represent a periodic discrete-time signal as a sum of complex exponential functions. DFS is a special case of Discrete Fourier Transform (DFT) that is used specifically for periodic signals.

The DFS of a periodic discrete-time signal x(n) with period N is given by:

x(n) = (1/N) * sum over k from 0 to N-1 (X(k) * e^(j*2*pi*k*n/N))

where X(k) is the discrete Fourier series coefficients of x(n) and is given by:

X(k) = sum over n from 0 to N-1 (x(n) * e^(-j*2*pi*k*n/N))

The DFS coefficients X(k) represent the amplitude and phase of each harmonic component of the periodic signal. The first DFS coefficient, X(0), represents the DC component of the signal, while the remaining coefficients represent the harmonics of the signal at integer multiples of the fundamental frequency.

DFS is useful in analysing periodic signals and identifying their harmonic components. The DFS coefficients can be used to compute the power spectral density of the periodic signal and to determine the frequency content of the signal. DFS is also used in signal processing applications such as digital audio and video compression, where it can be used to identify and remove redundant or irrelevant spectral components of a signal.

Explain Multiplication of 2DFT’s

Multiplication of two discrete Fourier transforms (DFTs) is a mathematical operation used in digital signal processing to obtain the frequency-domain representation of the product of two signals.

Suppose we have two discrete-time signals x(n) and y(n) of length N, whose DFTs are given by X(k) and Y(k), respectively. The product of x(n) and y(n) is given by:

z(n) = x(n) * y(n)

The DFT of the product z(n) can be computed by multiplying the DFTs of x(n) and y(n) at each frequency index k, i.e.,

Z(k) = X(k) * Y(k)

where Z(k) is the DFT of z(n).

In other words, the multiplication of two DFTs corresponds to a pointwise multiplication of their values in the frequency domain. This operation is also known as the frequency-domain convolution, or simply the Hadamard product.

Multiplication of DFTs finds applications in various areas of signal processing, such as filter design, signal analysis, and image processing. For example, multiplying the DFT of a signal with the DFT of a filter in the frequency domain can be used to implement a filter in a computationally efficient manner. In image processing, multiplying the DFTs of two images can be used to perform pointwise multiplication of their pixel values, known as image masking. This operation is used, for example, to blend two images or to remove selected regions from an image.

Describe Circular Convolution

Circular convolution is a type of convolution that is defined for finite-length sequences, where the end of the sequence is assumed to be connected to the beginning in a circular manner.

Mathematically, circular convolution of two sequences x(n) and h(n) of length N is defined as:

y(n) = sum over k from 0 to N-1 (x(k) * h((n-k) modulo N))

where modulo N denotes the modulo operation with respect to N.

Circular convolution can be computed efficiently using the Discrete Fourier Transform (DFT) and inverse DFT. Specifically, circular convolution can be expressed as the product of the DFTs of the sequences x(n) and h(n), i.e.,

Y(k) = X(k) * H(k)

Where X(k) and H(k) are the DFTs of x(n) and h(n), respectively, and Y(k) is the DFT of the circular convolution y(n).

The inverse DFT of Y(k) can be computed to obtain the circular convolution y(n) directly.

Circular convolution is commonly used in digital signal processing applications such as filtering and signal analysis. In particular, circular convolution is used to implement linear time-invariant (LTI) systems with finite-length impulse responses, such as digital filters. By padding the impulse response with zeros, circular convolution can also be used to implement LTI systems with longer impulse responses than the input signal. Circular convolution is also used in cyclic signal processing applications such as audio and video compression, where it is used to compress and decompress signals with periodic structures.