The B.Tech. AKTU Quantum Book will help you master the skill of digital signal processing. Gain access to crucial notes, frequent queries, and insightful knowledge in this area. Unit-5 Multirate Digital Signal Processing (MDSP)
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Q1. What is multirate digital signal processing ? Discuss about application areas of it.
Ans. A. Multirate digital signal processing:
- 1. Sampling rate conversion refers to the process of transforming a signal from one rate to another. Multirate digital signal processing systems are those that use several sample rates to process digital signals.
- 2. With an upsampler and a downsampler, different sampling rates can be achieved. To accomplish this, the basic procedures of multirate processing are decimation and interpolation.
- 3. Decimation is used to reduce the sample rate, whereas interpolation is used to increase the sampling rate.
B. Applications:
- i. Sub-band coding (speech, image)
- ii. Voice privacy using analog phone lines
- iii. Signal compression by subsampling
- iv. A/D, D/A converters.
C. Application area of multirate signal processing:
There are various areas in which multirate signal processing is used:
- i. Communication systems
- ii. Speech and audio processing systems
- iii. Antenna systems, and
- iv. Radar systems
Q2. Explain subband coding of speech signals.
Ans. 1. Subband coding is a method, where the speech signal is subdivided into several frequency bands and each band is digitally encoded separately.
![Explain subband coding of speech signals.](https://bachelorexam.com/wp-content/uploads/2023/10/image-398.png)
![Explain subband coding of speech signals.](https://bachelorexam.com/wp-content/uploads/2023/10/image-398.png)
- 2. Filter design is crucial in getting good subband performance coding. Aliasing caused by decimation of subband signals must be minimized.
- 3. Using Quadrature Mirror Filters (QMF) with the frequency response characteristics indicated in Fig. 4 is a feasible solution to the aliasing problem. The subband encoded speech signal generation approach is essentially the inverse of the encoding procedure.
- 5. As illustrated in Fig. 5, signals in nearby low-pass and high-pass frequency regions are interpolated, filtered, and merged. For each octave of the signal, a pair of (QMF) is employed in signal synthesis.
- 6 Subband coding is also an effective method to achieve data compression in image signal processing by combining subband coding with vector quantization for each subband signal.
![Explain subband coding of speech signals. Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-397.png)
![Explain subband coding of speech signals. Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-397.png)
- 7. When the signal energy is concentrated in a certain section of the frequency band, subband coding of signals is an effective way for accomplishing bandwidth compression in a digital representation of the signal.
Q3. Obtain the two-fold expanded signal y(n) of the input signal x(n)
![Obtain the two-fold expanded signal y(n) of the input signal x(n)](https://bachelorexam.com/wp-content/uploads/2023/10/image-402.png)
![Obtain the two-fold expanded signal y(n) of the input signal x(n)](https://bachelorexam.com/wp-content/uploads/2023/10/image-402.png)
Ans. 1. The output signal y(n) is given by
![Obtain the two-fold expanded signal y(n) of the input signal x(n) Btech Aktu](https://bachelorexam.com/wp-content/uploads/2023/10/image-401.png)
![Obtain the two-fold expanded signal y(n) of the input signal x(n) Btech Aktu](https://bachelorexam.com/wp-content/uploads/2023/10/image-401.png)
2. In general, to obtain the expanded signal y(n) by a factor I, (I- 1) zeros are inserted between the samples of the original signal x(n).
3. The z-transform of the expanded signal is
![Obtain the two-fold expanded signal y(n) of the input signal x(n) Aktu](https://bachelorexam.com/wp-content/uploads/2023/10/image-400.png)
![Obtain the two-fold expanded signal y(n) of the input signal x(n) Aktu](https://bachelorexam.com/wp-content/uploads/2023/10/image-400.png)
4. The input and output signals are shown in Fig.
![Obtain the two-fold expanded signal y(n) of the input signal x(n) Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-399.png)
![Obtain the two-fold expanded signal y(n) of the input signal x(n) Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-399.png)
Q4. Discuss modulation free method for decimation and interpolation.
Ans. A. Modulation free method for decimation:
1. Decimation of the sampled bandpass signal whose spectrum is shown in Fig. Note that the signal spectrum is confined to the frequency range,
![Discuss modulation free method for decimation and interpolation. Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-404.png)
![Discuss modulation free method for decimation and interpolation. Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-404.png)
2. A bandpass filter would normally be used to eliminate signal components outside the frequency desired of the frequency range. Then direct decimation bandpass signal by the factor D results in the Fig. spectrum shown in 5.8.2(a), for m odd, and Fig. 5.8.2(b) for m even.
![Discuss modulation free method for decimation and interpolation. Aktu Btech](https://bachelorexam.com/wp-content/uploads/2023/10/image-403.png)
![Discuss modulation free method for decimation and interpolation. Aktu Btech](https://bachelorexam.com/wp-content/uploads/2023/10/image-403.png)
3. In the case where m is odd, there is an inversion of the spectrum of the signal. This inversion can be undone by multiplying each sample of the decimation signal by (-1)n, n = 0, 1, …
B. Modulation free method for interpolation:
- 1 Modulation free interpolation of a bandpass signal by an integer factor I can be accomplished in a similar manner. The process of upsampling by inserting zeros between samples of x(n) produces I image in the band 0 ≤ ω ≤ 𝝅. The desired image can be selected by bandpass filtering.
- 2. Note that the process of interpolation also provides us with the opportunity to achieve frequency translation of the spectrum.
Q5. A one stage decimator is characterized by the following :
Decimation factor = 3
Anti-aliasing filter coefficients
![A one stage decimator is characterized by the following :](https://bachelorexam.com/wp-content/uploads/2023/10/image-410.png)
![A one stage decimator is characterized by the following :](https://bachelorexam.com/wp-content/uploads/2023/10/image-410.png)
Given the data, r(n) with successive values [6, – 2, – 3, 8, 6, 4,- 2], calculate and list the filtered output w(n) and the output of the decimator y(m).
Ans.
![A one stage decimator is characterized by the following : Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-409.png)
![A one stage decimator is characterized by the following : Digital Signal Processing](https://bachelorexam.com/wp-content/uploads/2023/10/image-409.png)
![A one stage decimator is characterized by the following : Aktu](https://bachelorexam.com/wp-content/uploads/2023/10/image-408.png)
![A one stage decimator is characterized by the following : Aktu](https://bachelorexam.com/wp-content/uploads/2023/10/image-408.png)
Filtered output:
![A one stage decimator is characterized by the following : Btech](https://bachelorexam.com/wp-content/uploads/2023/10/image-407.png)
![A one stage decimator is characterized by the following : Btech](https://bachelorexam.com/wp-content/uploads/2023/10/image-407.png)
![A one stage decimator is characterized by the following : Aktu Btech](https://bachelorexam.com/wp-content/uploads/2023/10/image-406.png)
![A one stage decimator is characterized by the following : Aktu Btech](https://bachelorexam.com/wp-content/uploads/2023/10/image-406.png)
Decimator output :
![A one stage decimator is characterized by the following : Digital](https://bachelorexam.com/wp-content/uploads/2023/10/image-405.png)
![A one stage decimator is characterized by the following : Digital](https://bachelorexam.com/wp-content/uploads/2023/10/image-405.png)
Q6. Suppose that the input signal has a flat spectrum between – 0.8𝝅 and 0.8𝝅. Determine the number of subfilters to achieve a signal to distortion ratio of 50 dB when linear interpolation is applied.
Ans.
![bachelor exam preparation all question paper with solution important questions with solution](http://bachelorexam.com/wp-content/uploads/2023/01/cute-teddy1.png)
![bachelor exam preparation all question paper with solution important questions with solution](http://bachelorexam.com/wp-content/uploads/2023/01/cute-teddy1.png)
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