ISSN: 2320-5407 Int. J. Adv. Res.

13(02), 763-773

Journal Homepage: - www.journalijar.com

Article DOI: 10.21474/IJAR01/20434

DOI URL: http://dx.doi.org/10.21474/IJAR01/20434

RESEARCH ARTICLE
VOCAL CHAMELEON

Vivek Raviraj D.1, Sakshi Shivakumar1, Lakshmi Bhaskar2 and Kiran K.N3
1. Student, Department of Electronics and Communication, BNM Intitute of Technology, Bengaluru.
2. Associate Professor, Department of Electronics and Communication, BNM Intitute of Technology, Bengaluru.
3. Assistant Professor, Department of Electronics and Communication, BNM Intitute of Technology, Bengaluru.
……………………………………………………………………………………………………....
Manuscript Info Abstract
……………………. ………………………………………………………………
Manuscript History To revolutionize the karaoke experience, this work proposes the
Received: 14 December 2024 development of a sophisticated system for processing vocal files and
Final Accepted: 17 January 2025 seamlessly merging them with high-quality karaoke accompaniments.
Published: February 2025 Traditional karaoke tracks often lack the personal touch and
professional quality that the users desire. This project addresses these
Key words:-
Vocals, Karaoke, Equalization, Reverb, issues by implementing advanced vocal effects and optimizing
Pitch Correction, Chorus, Phase performance to enhance accuracy. Key objectives include integrating
Vocoder, Audio Enhancement, Spectral features like equalization, pitch correction, normalization, reverb and
Analysis, Wavelet Transform
chorus and optimizing algorithms for efficient processing, and enabling
collaborative features for users to work together and share their
creations. The project targets to create a versatile and robust tool that
meets the needs of a global audience. The result is a user-friendly
application that empowers users to create personalized and professional
sounding karaoke tracks, enhancing both personal enjoyment and
professional music production. This innovative approach to vocal
processing and merging paves the way for new possibilities in the
world of digital music and entertainment.
Copyright, IJAR, 2025,. All rights reserved.
……………………………………………………………………………………………………....
Introduction:-
The advancement of digital signal processing (DSP) technology has revolutionized the field of audio processing,
enabling sophisticated manipulation and analysis of sound signals. This project report delves into the realm of audio
processing, with a particular focus on the processing of vocal files using MATLAB, a high-performance language and
environment for technical computing. In recent years, audio processing has gained significant traction in various
applications, ranging from music production and speech recognition to telecommunications and hearing aids. The
ability to enhance, filter, and analyse vocal recordings holds immense potential in improving the clarity, quality, and
intelligibility of speech signals. This is particularly important in scenarios such as noise reduction in
telecommunication systems, automatic transcription in speech-to-text applications, and the enhancement of audio
quality in media production. The goal of the project is to enhance vocal recordings by applying post-processing
techniques to the vocals and then mix them with a karaoke accompaniment. The project employs MATLAB to
develop and implement algorithms for the processing of vocal files. An additional karaoke file is included to which
the vocal file will be added. The processed vocal file and the karaoke file are merged at the end to give a soothing and
euphonious audio overall. MATLAB, with its extensive library of built-in functions and toolboxes, provides a robust
platform for audio signal analysis and manipulation. The project will cover various aspects of audio processing,
including normalization, equalization, pitch correction, addition of reverb and chorus. The project provides an

Corresponding Author:- Vivek Raviraj D. 763

Address:- Student, Department of Electronics and Communication, BNM Intitute of
Technology, Bengaluru.
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overview of the fundamental concepts in audio signal processing, highlighting the key challenges and techniques
involved. It discusses the specific requirements and objectives of our project, followed by a detailed description of the
methodologies employed in processing the vocal files. Subsequently, the project presents the results obtained from the
implemented algorithms, demonstrating the effectiveness of our approach through quantitative and qualitative
analysis. Finally, it concludes the report with a discussion of the findings, potential applications, and future directions
for further research and development in this field.

Literature Survey:-
A comprehensive overview of the fundamental concepts and advanced techniques in the processing of speech and
audio signals. Published in 2016, this work delves into both theoretical and practical aspects, addressing essential
topics such as signal representation, feature extraction, and various processing algorithms. The book is notable for
its balanced treatment of traditional approaches alongside emerging trends, offering insights into the development of
robust and efficient systems for applications ranging from speech recognition and synthesis to audio enhancement
and compression. McLoughlin emphasizes the importance of understanding the underlying physical and perceptual
properties of audio signals, providing a strong foundation for further research and development in the field [1].

Signal processing technology - A detailed exploration of audio signal processing techniques with a focus on practical
implementation using MATLAB. This work is particularly valuable for its hands-on approach, guiding readers
through a series of experiments and projects that demonstrate key concepts and algorithms in audio processing.
Topics covered include digital signal processing basics, filter design, time-frequency analysis, and various
applications in noise reduction, echo cancellation, and audio effects. The use of MATLAB as a tool for simulation
and analysis enables readers to visualize the effects of different processing techniques and gain a deeper
understanding of their practical implications. This book serves as both a textbook for students and a reference for
practitioners in the field. The book covers a wide range of topics, including continuous-time and discrete-time signals,
linear time-invariant systems, Fourier analysis, and Laplace and Z-transform techniques. Hsu's clear and systematic
approach makes complex concepts accessible, with numerous examples and exercises to reinforce understanding.
This work is essential for anyone studying or working in fields that require a solid grasp of signal processing
principles, such as electrical engineering, communications, and control systems. The theoretical foundations laid out
in this book underpin many of the advanced techniques discussed in more specialized audio and speech processing
literature [2,3].

This work is particularly relevant in the context of telecommunications and digital communication systems, where
bandwidth efficiency and speech intelligibility are critical. Paliwal explores various coding techniques, including
linear predictive coding (LPC), code-excited linear prediction (CELP), and other advanced methods that balance
compression efficiency with perceptual quality. The book also delves into speech synthesis techniques, highlighting
the interplay between naturalness and intelligibility in synthetic speech. By providing a detailed examination of both
coding and synthesis, this work offers valuable insights into the design and implementation of modern speech
processing systems [4].

Wavelets provide a multi-resolution analysis framework that is particularly suited for analysing nonstationary signals,
making them ideal for applications in both audio and image processing. Morgan's work covers the mathematical
foundations of wavelets, various wavelet transform techniques, and their applications in de-noising, compression, and
feature extraction. The book highlights the advantages of wavelet-based methods over traditional Fourier-based
approaches, particularly in handling signals withLocalized time-frequency characteristics. This work is instrumental
for researchers and practitioners seeking to leverage wavelet techniques for advanced signal and image processing
tasks. In conclusion, these works collectively represent a broad spectrum of research and practical advancements in
the field of audio and speech processing. From foundational theories and algorithms to practical implementations and
emerging technologies, they provide a rich resource for understanding and innovating in this dynamic field [5].

Early work in DSP focused on real-time processing for effects such as reverberation, echo, pitch shifting,
equalization, and distortion, with researchers like J. Moorer (1979) and Zölzer (2002) contributing to core algorithms.
Key developments include the phase vocoder for pitch shifting, wave-shaping for distortion, and FIR/IIR filters for
equalization. Real-time processing remains a significant challenge, especially in live applications, and ongoing
research, such as by Valimaki (2000) and Zölzer (2012), aims to optimize DSP algorithms for low-latency, high-
quality audio effects. Sharma and Prabhu's work builds on these foundations, focusing on more efficient real-time
implementations of sound effects in modern audio systems [6].

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Pitch detection algorithms are crucial for various applications, such as music analysis, speech processing, and audio
synthesis. The study provides a comprehensive comparison of different pitch detection methods, evaluating their
performance in terms of accuracy, computational complexity, and robustness. The authors systematically analyze
several algorithms, including time-domain methods, frequency-domain approaches, and hybrid techniques. Their
work highlights the strengths and weaknesses of each algorithm, offering insights into their suitability for specific
applications. By examining factors like algorithmic efficiency and reliability under varying conditions, the paper
contributes to a deeper understanding of pitch detection and aids in selecting appropriate techniques for different
practical scenarios. This comparative analysis serves as a valuable resource for researchers and practitioners aiming to
implement or improve pitch detection systems in their projects [7].

Methodology:-
To enhance vocal recordings for karaoke, the process begins with data acquisition by obtaining high-quality vocal
recordings and a karaoke track for accompaniment. In the pre-processing stage, the vocal recording is normalized to
ensure consistent volume levels. Audio processing techniques include equalization to adjust frequency balances,
pitch correction to maintain proper tuning, reverb addition for depth and space, and a chorus effect to enrich the
vocal sound. During mixing, time alignment ensures the vocal is synchronized with the karaoke track, followed by
volume balancing to achieve a harmonious blend. The vocal and karaoke tracks are then merged to create the final
mixed audio. Post-processing involves final normalization for consistent volume and exporting the audio in the
desired format, such as WAV or MP3.

Block Diagram.

Fig 3.1:- Vocal Processing.

Figure 3.1 depicts a straightforward block diagram illustrating the process of vocal processing. The process begins
with the Input Audio Signal (Vocals), which is the raw audio containing the vocals that need processing. This signal
enters the system and moves to the Vocal Processing stage, the core component of the system where the audio
undergoes various processing techniques. These techniques can include pitch shifting, equalization, compression,
and other audio effects designed to enhance or modify the vocals. After processing, the enhanced or modified audio
emerges as Processed Vocals, representing the final output. The arrows in the diagram indicate the directional flow
of the audio signal from its initial raw state to its processed form.

Fig 3.2:- Equalization.

Figure 3.2 shows the text "Input Audio" followed by "Signal (Vocals)" and "Equalized Vocals" written in a bold,
capitalized font. The background is a light, neutral color that contrasts with the black text. Overall, the image
appears to represent a graphic or visual representation of an audio processing system or software. It seems to depict
the process of inputting audio, equalizing the vocals, and potentially showcasing the before-and-after effects of
equalization. This visual representation could be used in audio production, sound engineering, or music editing to
demonstrate the signal processing stages of vocal equalization. The use of bold and capitalized text in a simple
layout makes the Information easily comprehensible, fitting for instructional or informational purposes within the
audio industry.

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Fig 3.3:- Chorus Effect.

Figure 3.3 displays a visual representation of the steps involved in processing audio vocals. At the top, the text
"Input Audio" is prominently featured, indicating the initial step of receiving audio input. Below that, the phrase
"Signal (Vocals)" suggests the focus on processing vocal signals. The subsequent phrase "Chorus Effect" denotes
the specific audio effect being applied to the vocals. The design features a clean layout with the text in a bold,
capitalized font, set against a light, neutral background, emphasizing clarity and simplicity. This graphic could be
employed in audio production or sound editing contexts to illustrate the workflow of applying a chorus effect to
vocals, providing a visual guide for users in the audio processing field.

Fig 3.4:- Reverb Effect.

Figure 3.4 depicts a visualization related to audio processing, specifically showing input audio vocals with a signal
(vocals) reverb effect applied. The graphical representation likely illustrates the waveform or spectral analysis of the
audio input, showcasing a distinctive pattern that reflects the presence of vocals. The inclusion of a reverb effect
suggests that the audio signal has been modified to simulate the reverberation or echo effect commonly used in
audio production to add depth and spatial realism to the sound. The visualization may feature various peaks and
valleys in the waveform, Indicating the varying intensity and frequency components of the vocals with the applied
reverb effect. Overall, the image captures a snapshot of the audio processing technique that enhances the texture and
ambiance of the vocals within the sound production.

Fig 3.5:- Delay Effect.

Figure 3.5 consists of a visual representation of an audio processing scenario. It illustrates the manipulation of input
audio vocals with a signal (vocals) delay effect. The graph likely portrays the waveform or spectral analysis of the
audio, depicting the characteristics of the vocals along with the application of the delay effect. The delay effect,
commonly used in audio production, involves creating echoes or repetitions of the original sound, altering the
perception of time and creating a sense of spaciousness. Consequently, the visualization may exhibit repeated
patterns in the waveform, symbolizing the delayed audio signals. This image captures a moment in the audio
processing chain, showcasing the transformation of the vocal input through the deliberate application of the delay
effect, providing a nuanced and time-altering dimension to the sound.

Fig 3.6:- Pitch Correction.

Figure 3.6 features a visual representation of an audio waveform, showcasing the intricate patterns and fluctuations
that characterize the pitch-corrected vocals. The graph appears to illustrate the precise adjustments made to the
pitch, highlighting the meticulous process of fine-tuning the vocal performance. The tag "(Vocals)" stands out,

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signaling the significance of this element within the audio content. Evidently, this image provides insight into the
technical manipulation of audio signals to enhance the quality and accuracy of vocal recordings, offering a glimpse
into the meticulous work involved in the production and refinement of music and other audio projects.

Alogorithm
START
STEP 1: Load the Audio Files:
 Read the audio files for vocals and karaoke using an appropriate function.

STEP 2: Check Sampling Rates:

 Ensure that the sampling rates of both the vocals and karaoke audio files are the same.
 If they differ, raise an error indicating the mismatch.

STEP 3: Normalize Audio:

 Normalize both the vocals and karaoke audio to prevent clipping by dividing each signal by its maximum
absolute value.

STEP 4: Apply Reverb Effect to Vocals:

 Define a gain for the reverb effect.
 Calculate the reverb delay based on a specified time and the sampling rate.
 Add the delayed and scaled version of the vocals to the original vocals to create the reverb effect.
 Normalize the reverb-processed vocals again

STEP 5: Apply Equalization:

 Define the frequency and gain for the equalization (EQ) to boost high frequencies.
 Design a parametric equalizer filter using the specified frequency, gain, and sampling rate.
 Apply the EQ filter to the reverb-processed vocals.

STEP 6: Apply Chorus Effect:

 Define the depth and rate for the chorus effect.
 For each sample in the equalized vocals, calculate a delayed index using a sinusoidal modulation.
 If the delayed index is within the valid range, mix the original and delayed samples to create the chorus effect.

STEP 7: Apply Delay Effect:

 Define the gain and delay time for the delay effect.
 Add the delayed and scaled version of the chorus-processed vocals to the original vocals.
 Normalize the delay-processed vocals again.

STEP 8: Plot Waveforms:

 Plot the waveforms of the vocals at various stages: original, with reverb, equalized, with chorus, and with delay.
 Plot the waveform of the karaoke audio.

STEP 9: Mix Vocals with Karaoke:

 Determine the minimum length between the processed vocals and karaoke.
 Overlap and mix the vocals and karaoke by adding them together for the duration of the shorter length.
 Normalize the mixed audio to prevent clipping.

STEP 10: Save the Mixed Audio:

 Write the mixed audio to a new file.

STEP 11: Plot Final Mixed Audio:

 Plot the waveform of the final mixed audio.

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STEP 12: Optional: Plot Spectrograms:

 Plot the spectrograms of the original vocals, processed vocals, and karaoke for visual analysis.

STEP 13: Play the Final Mixed Audio:

 Optionally, play the final mixed audio using an appropriate sound playback function.

STEP 14: Design Parametric Equalizer Function:

 Design a parametric equalizer to boost frequencies around a specified frequency by a specified gain.
 Calculate filter coefficients based on the frequency, gain, quality factor, and sampling rate.
 Normalize the filter coefficients.

END

Mathematical formulas
1.Normalization: Normalization scales audio samples so their amplitude lies within a specific range, typically[-1,
1]. This prevents clipping and ensures a consistent volume.
If x is the input signal, normalization is defined as:
Xnorm = X/(max(|X|)…….(1)
2. Reverb: Reverb simulates the persistence of sound in a space by adding a delayed and scaled version of the
signal.
Add a delayed version of the signal to itself:
Y[n] = X[n] + G⋅X [n−D]…….(2)
Where G is reverb gain, D is delay in samples.
3. Equalization: Equalizationmodifies the frequency content of a signal. A parametric EQ boosts or attenuates
specific frequency bands.

A parametric EQ is implemented using a second-order digital filter defined by its transfer function:

H (z) = (b0 + b1z^(−1) + b2z^(−2)) / (a0 + a1z^(−1) + a2z^(−2))…….(3)

Where b and a are the filter coefficients.
Coefficients calculation in designParamEQ:
Amplitude Factor: A=10 ^ (gain / 40)
Center Frequency: w0 = 2phi (freq / fs)
Bandwidth: Q=1

4. Delay effect: Delay introduces an echo by adding a scaled and time-shifted version of the original signal.

Y[n] = X[n] + G⋅X [n−D]…….(4)

5. Mixing Audio: Mixing involves summing two audio signals after ensuring they have the same length and are
normalized.

If x1[n] and x2[n] are two signals:

y[n] = x1[n] + x2[n]…….(5)

6. Wave form plotting: Waveforms visualize the audio signals in the time domain.For an audio signal x with N
samples, plot:X-axis: Sample indices (1 to N) and Y-axis: x[n].

Results:-
The results of our project on audio processing for vocal files using MATLAB demonstrate significant improvements
across multiple stages of the processing pipeline, including normalization, equalization, pitch correction, and
spectral analysis. Equalization was applied to adjust the frequency components of the vocal recordings, improving
overall quality. Various filters, such as low-pass, high-pass, and band-pass filters, were designed and implemented.
The low-pass filters effectively reduced high-frequency noise and hiss without significantly affecting vocal quality,

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while the high-pass filter eliminated low-frequency hum and rumble, making the vocals clearer. The band-pass filter
enhanced mid-range frequencies crucial for speech intelligibility. Spectrogram analysis before and after equalization
showed a more balanced frequency distribution. Pitch correction, vital for tuning vocal recordings, used the
autocorrelation method. Quantitative evaluation showed a reduction in pitch error from ±50 cents to ±5 cents using
the phase vocoder, leading to harmonically balanced and in-tune recordings. Normalization adjusted the amplitude
of the vocal recordings to a consistent level, ensuring uniformity and preventing distortion, with increased overall
loudness making softer parts more audible without compromising integrity. The chorus effect added depth by
introducing delayed and pitch-modulated copies of the signal, simulating the effect of multiple voices, enhancing
stereo imaging, and creating a richer sound. Reverb simulated different acoustic environments, adjusting parameters
like room size and decay time to create spatial depth and realism, enhancing the naturalness of the vocals.
Spectrogram analysis visually confirmed these improvements, showing a cleaner, more defined spectral
representation, and both quantitative and qualitative evaluations indicated significant enhancements in the processed
audio.

Fig 5.1:- Spectrogram.

Figure 5.1displaysthree spectrograms,eachrepresentingdifferentstagesofaudioprocessing.Thefirst spectrogram,

labeled "Original Vocals," shows the power of the original vocal signal acrossfrequencies ranging from 0 to 20
kHz over a time span of 0 to 1 minute. The second spectrogram, labeled
"ProcessedVocals,"alsocoversthesamefrequencyandtimerangesbutillustratesthepowerof the vocal signal after
undergoing some form of processing, which could include noise reduction, equalization, or other audio effects. The
third spectrogram, labeled "Karaoke," represents the instrumental version of the audio, where the vocals have been
removed or significantly suppressed, makingitsuitableforkaraoke.Eachspectrogram usesacolor scalefrom
blue(indicatinglowpower)to yellow (indicating high power) to depict the intensity of the signal at various
frequencies and times.

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Fig 5.2:- Mixed audio.

Figure 5.2 shows a waveform plot labeled "Mixed Audio." The horizontal axis represents the sample number,
spanning from 0 to approximately 2.7 million samples, indicating a substantial duration of audio data. The vertical
axis represents the amplitude of the audio signal, ranging from -1 to 1. The plot illustrates the variations in
amplitude over the entire duration of the audio, with a densely packed waveform indicating a complex mixture of
sounds. The signal exhibits high amplitude variations throughout most of the duration, suggesting a dynamic audio
with potentially loud and quiet sections, peaks, and troughs. This type of plot is commonly used to visualize the raw
waveform of an audio signal, providing insight into its overall structure and dynamics.

Fig 5.3:- All audio signals

Figure 5.3 presents a series of six waveform plots, each depicting different stages of audio processing. The first plot,
labeled "Original Vocals," shows the unprocessed vocal track with amplitude variations indicating the dynamics of
the original recording. The second plot, "Vocals with Reverb," displays the vocal track after adding reverb, resulting

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in a smoother and more spacious sound, as evidenced by the less sharp and more blended peaks and troughs. The
third plot, "Equalized Vocals," represents the vocal track after equalization, which adjusts the balance of different
frequency components, leading to changes in the signal's tonal balance reflected in the amplitude variations. The
fourth plot, "Vocals with Chorus," shows the vocal track processed with a chorus effect, adding depth and richness
by simulating multiple voices, visible as a slightly more complex and thicker waveform. The fifth plot, "Vocals with
Delay," illustrates the vocal track with a delay effect, characterized by repeated echoes, which might show repeating
patterns or extended tails on peaks. The final plot, labeled "Karaoke," depicts the instrumental version of the audio
with the vocals removed or significantly suppressed, resulting in a less dense waveform with distinct gaps,
indicating the absence of vocal components. Each plot uses a horizontal axis representing the sample number,
ranging from 0 to approximately 2.7 million samples, and a vertical axis for amplitude, highlighting the differences
in the audio processing techniques applied to the vocal track.

Fig 5.4:- Pitch Correction wave form.

Figure 5.4 displays a graph with two overlaid audio signals: one labeled „Original‟ and the other „Pitch Shifted.‟
The horizontal axis represents time (in seconds), ranging from 0 to 70 seconds. The vertical axis represents
amplitude (ranging from -0.6 to 0.6). The Original‟ signal is depicted in blue, while the “Pitch Shifted‟ signal is
shown in red. Both signals exhibit fluctuating waveforms, suggesting variations in pitch and volume over time. This
graph could be relevant for analyzing how pitch shifting affects an audio signal’s waveform and amplitude.

Fig 5.5:- UI for processing and mixing.

Figure 5.5 is the graphical user interface (GUI) for audio processing, likely developed using MATLAB, features
multiple waveform plots and control buttons for managing vocal and karaoke tracks. Users can load vocal and
karaoke files, apply various audio effects, play the processed audio, and save the final output. The interface includes
six waveform plots representing different stages of audio processing: original vocals, vocals with reverb, equalized
vocals, vocals with chorus, vocals with delay, and mixed audio. These plots display the audio signal's sample

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number versus amplitude, allowing users to visualize changes at each stage. The GUI's intuitive design, with buttons
like "Load Vocals," "Load Karaoke," "Apply Effects," "Play," and "Save," offers a streamlined experience for
loading, processing, and mixing audio tracks. It effectively caters to both amateur and professional audio engineers
by providing real-time feedback on audio adjustments and facilitating easy navigation through various audio effects.

Conclusion and Future Scope:-

In conclusion, the project on audio processing for vocal files and merging with karaoke accompaniment using
MATLAB has demonstrated significant advancements and achievements in enhancing the quality, clarity, and
artistic appeal of vocal recordings. Through a systematic methodology encompassing equalization, pitch correction,
normalization, chorus, and reverb effects, The project has successfully transformed raw vocal recordings into
smooth and specialized audio tracks. MATLAB proved to be an indispensable tool throughout the project, offering a
robust platform for implementing and optimizing complex audio processing algorithms.The integration of
MATLAB‟s Signal Processing Toolbox, Audio Toolbox, and custom scripting facilitated precise control and
thorough analysis at each processing stage. This empowered the project to achieve meticulous adjustments and
refinements, resulting in polished and professional-quality audio outputs. Looking forward, future avenues of
exploration could delve into advanced machine learning techniques for advanced techniques in dynamic range
compression, machine learning integration for intelligent audio processing, and real-time applications for interactive
audio experiences, automated parameter optimization and real-time audio processing applications. These
advancements aim to further elevate the capabilities and efficiency of audio processing technologies, expanding
their impact across diverse domains including music production, telecommunications, and interactive multimedia
experiences.

The project exemplifies the transformative potential of MATLAB-based audio processing techniques in enhancing
vocal recordings. By leveraging state-of-the-art methodologies and tools, the project has not only met but exceeded
its objectives, setting a foundation for continued innovation and excellence in the field of audio engineering and
production.

Acknowledgment:-
The satisfaction and euphoria that accompany the successful completion of the MATLAB Project would be
incomplete without mentioning the names of the people who made it possible. We express our heartfelt gratitude to
B N M Institute of Technology, for giving us the opportunity to pursue Degree of Electronics and Communication
Engineering and helping us to shape our career. We take this opportunity to thank Prof. T. J. Rama Murthy,
Director, BNMIT, Dr. S.Y Kulkarni, Additional Director and Principal, BNMIT, Prof. Eishwar N Maanay, Dean,
BNMIT and Dr. Krishnamurthy. G. N, Deputy Director, BNMIT for their support and encouragement to pursue
this project. We would like to thank Dr. Yasha Jyothi M Shirur, Professor and Head, Dept. of Electronics and
Communication Engineering, for her support and encouragement. It is with a deep sense of gratitude and great
respect; we owe our indebtedness to our guide Smt. Lakshmi Bhaskar, Assistant professor. We thank her for her
constant encouragement during the execution of this project. We thank our parents, without whom none of this
would have been possible. Their patience and blessings have been with us at every step of this Project. We would
express our thanks to all our friends and all those who have helped us directly or indirectly for the successful
completion of the Project.

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