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Audio Signal Processing and Coding

Andreas Spanias, Ted Painter, Venkatraman Atti

Chapter 2

SIGNAL PROCESSING ESSENTIALS - all with Video Answers

Educators


Chapter Questions

01:19

Problem 1

Determine the continuous Fourier transform (CFT) of a pulse described by
$$
x(t)=u(t+1)-u(t-1),
$$
where $u(t)$ is the unit step function.

Amit Srivastava
Amit Srivastava
Numerade Educator

Problem 2

State and derive the CFT properties of duality, time shift, modulation, and convolution.

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02:17

Problem 3

For the circuit shown in Figure 2.34(a) and for $\mathrm{RC}=1$,
a. Write the input-output differential equation.
b. Determine the impulse response in closed-form by solving the differential equation.
c. Write the frequency response function.
d. Determine the steady state response, $y(t)$, for $x(t)=\sin (10 t)$.
FIGURE CANT COPY
Figure 2.34. (a) A simple RC circuit; (b) input signal for problem 2.3(e), $x(t)$; and (c) input signal for problem 2.3(f).
e. Given $x(t)$ as shown in Figure 2.34(b), find the circuit output, $y(t)$, using convolution.
f. Determine the Fourier series of the output, $y(t)$, of the RC circuit for the input shown in Figure 2.34(c).

Mayukh Banik
Mayukh Banik
Numerade Educator

Problem 4

Determine the CFT of $p(t)=\sum_{n=-\infty}^{\infty} \delta\left(t-n T_s\right)$. Given, $x_s(t)=x(t) p(t)$, derive the following,
$$
\begin{aligned}
& X_s(\omega)=\frac{1}{T_s} \sum_{k=-\infty}^{\infty} X\left(\omega-k \omega_s\right) \\
& x(t)=\sum_{n=-\infty}^{\infty} x\left(n T_s\right) \operatorname{sinc}\left(B\left(t-n T_s\right)\right),
\end{aligned}
$$
where $X(\omega)$ and $X_s(\omega)$ are the spectra of ideally bandlimited and uniformly sampled signals, respectively, and $\omega_s=2 \pi / T_s$. (Refer to Figure 2.8 for variable definitions.)

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Problem 5

Determine the $z$-transforms of the following causal signals:
a. $\sin (\Omega n)$
b. $\delta(n)+\delta(n-1)$
c. $p^n \sin (\Omega n)$
d. $u(n)-u(n-9)$

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Problem 6

Determine the impulse and frequency responses of the averaging filter
$$
h(n)=\frac{1}{L+1}, n=0,1, \ldots, L, \text { for } L=9 .
$$

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Problem 7

Show that the IDFT can be derived as a least squares signal matching problem, where $N$ points in the time domain are matched by a linear combination of $N$ sampled complex sinusoids.

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Problem 8

Given the transfer function $H(z)=(z-1)^2 /\left(z^2+0.81\right)$,
a. Determine the impulse response, $h(n)$.
b. Determine the steady state response due to the sinusoid $\sin \left(\frac{\pi n}{4}\right)$.

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Problem 9

Derive the decimation-in-time FFT algorithm and determine the number of complex multiplications required for an FFT size of $N=1024$.

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03:56

Problem 10

Derive the following expression in a simple two-band QMF
$$
X_{d, k}\left(e^{j \Omega}\right)=\frac{1}{2}\left(X_k\left(e^{j \Omega / 2}\right)+X_k\left(e^{j(\Omega-2 \pi) / 2}\right)\right), \quad k=0,1 .
$$

Refer to Figure 2.28 for variable definitions. Give and justify the conditions for alias-free reconstruction in a simple QMF bank.

Mihajlo Grcic
Mihajlo Grcic
Numerade Educator
04:18

Problem 11

Design a tree-structured uniform QMF bank that will divide the spectrum of $0-20 \mathrm{kHz}$ into eight uniform subbands. Give appropriate figures and denote on the branches the range of frequencies.

Amit Srivastava
Amit Srivastava
Numerade Educator
04:18

Problem 12

Modify your design in problem 2.11 and give one possible realization of a simple nonuniform tree structured QMF bank that will divide the of 0$20 \mathrm{kHz}$ spectrum into eight subbands whose bandwidth increases with the center frequency.

Amit Srivastava
Amit Srivastava
Numerade Educator
04:18

Problem 13

Design a low-pass shelving filter for the following specifications: $f_s=$ $16 \mathrm{kHz}, f_c=4 \mathrm{kHz}$, and $g=10 \mathrm{~dB}$.

Amit Srivastava
Amit Srivastava
Numerade Educator
02:34

Problem 14

Design peaking filters for the following cases: a) $\Omega_c=\pi / 4, Q=2, g=$ $10 \mathrm{~dB}$ and b) $\Omega_c=\pi / 2, Q=3, g=5 \mathrm{~dB}$. Give frequency responses of the designed peaking filters.

Amit Srivastava
Amit Srivastava
Numerade Educator

Problem 15

Design a five-band digital audio equalizer using the concept of peaking digital filters. Select center frequencies, $f_c$, as follows: $500 \mathrm{~Hz}, 1500 \mathrm{~Hz}$, $4000 \mathrm{~Hz}, 10 \mathrm{kHz}$, and $16 \mathrm{kHz}$; sampling frequency, $f_s=44.1 \mathrm{kHz}$ and the corresponding peaking filter gains as $10 \mathrm{~dB}$. Choose a constant $\mathrm{Q}$ for all the peaking filters.

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01:44

Problem 16

Derive equations (2.63) and (2.64).

Narayan Hari
Narayan Hari
Numerade Educator

Problem 17

Derive equation (2.66).

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Problem 18

Show that the PSD is real-valued and positive.

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01:44

Problem 19

Show that the estimator (2.68) of the autocorrelation is biased.

Rashmi Sinha
Rashmi Sinha
Numerade Educator
01:44

Problem 20

Show that the estimator (2.68) provides autocorrelations such that $r_{x x}(0) \geqslant$ $\left|r_{x x}(m)\right|$.

Rashmi Sinha
Rashmi Sinha
Numerade Educator
07:57

Problem 21

Provide an unbiased autocorrelation estimator by modifying the estimator in (2.68).

Abhirup Pal
Abhirup Pal
Numerade Educator

Problem 22

A digital filter with impulse response $h(n)=0.7^n u(n)$ is excited by white Gaussian noise of zero mean and unit variance. Determine the mean and variance of the output of the digital filter in closed-form during the transient and steady state.

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Problem 23

The filter $H(z)=z /(z-0.8)$ is excited by white noise of zero mean and unit variance.
a. Determine all the autocorrelation values at the output of the filter at steady state.
b. Determine the PSD at the output.

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Problem 24

Consider the 2-band QMF bank shown in Figure 2.35. In this figure, $x(n)$ denotes speech frames of 256 samples and $\hat{x}(n)$ denotes the synthesized speech frames.
a. Design the transfer functions, $F_0(z)$ and $F_1(z)$, such that aliasing is cancelled. Also calculate the overall delay of the QMF bank.
b. Select an arbitrary voiced speech frame from $\mathrm{Ch} 2$ speech.wav. Give time-domain and frequency-domain plots of $x_{d 0}(n)$ and $x_{d 1}(n)$ for that particular frame. Comment on the frequency-domain plots with regard to low-pass/high-pass band-splitting.
Figure 2.35. A two-band QMF bank.
FIGURE CANT COPY
Figure 2.36. Speech synthesis from a select number (subset) of FFT components.
FIGURE CANT COPY
Table 2.2. Signal-to-noise ratio (SNR) and MOS values.
$$
\begin{array}{|c|c|c|}
\hline \begin{array}{l}
\text { Number of FFT } \\
\text { components, } L
\end{array} & \mathrm{SNR}_{\text {overall }} & \begin{array}{l}
\text { Subjective evaluation MOS (mean } \\
\text { opinion score) in a scale of } \\
1-5 \text { for the entire speech record }
\end{array} \\
\hline \begin{array}{l}
16 \\
128
\end{array} & & \\
\hline
\end{array}
$$
c. Repeat step (b) for $x_1(n)$ and $x_{d 1}(n)$ in order to compare the signals before and after the downsampling stage.
d. Calculate the SNR between the input speech record, $x(n)$, and the synthesized speech record, $\hat{x}(n)$. Use the following equation to compute the SNR,
$$
\mathrm{SNR}=10 \log _{10}\left(\frac{\sum_n x^2(n)}{\sum_n(x(n)-\hat{x}(n))^2}\right)(\mathrm{dB})
$$

Listen to the synthesized speech record and comment on its quality.
e. Choose a low-pass $F_0(z)$ and a high-pass $F_1(z)$, such that aliasing occurs. Compute the SNR. Listen to the synthesized speech and describe its perceptual quality relative to the output speech in step(d). (Hint: Use first-order IIR filters.)

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Problem 25

In Figure 2.36, $x(n)$ denotes speech frames of 256 samples and $x^{\prime}(n)$ denotes the synthesized speech frames. For $L=N(=256)$, the synthesized speech will be identical to the input speech. In this computer exercise, you need to perform speech synthesis on a frame-by-frame basis from a select number (subset) of FFT components, i.e., $L<N$. We will use two methods for the FFT component selection, (i) Method 1: selecting the first $L$ components including their conjugate-symmetric ones out of a total of $N$ components; and (ii) Method 2: the least-squares method (peak-picking method that selects the $L$ components that minimize the sum of squares error.)
a. Use $L=64$ and Method 1 for component selection. Perform speech synthesis and give time-domain plots of both input and output speech records.
b. Repeat the above step using the peak-picking Method 2 (choose $L$ peaks including symmetric components in the FFT magnitude spectrum). List the SNR values in both the cases. Listen to the output files corresponding to (a) and (b) and provide a subjective evaluation (on a MOS scale 1-5). To calibrate the process think of a wireline telephone quality (toll) as 4, cellphone quality around 3.7.
c. Perform speech synthesis for (i) $L=16$ and (ii) $L=128$. Use the peakpicking Method 2 for the FFT component selection. Compute the overall SNR values and provide a subjective evaluation of the output speech for both the cases. Tabulate your results in Table 2.2.

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