1. The document describes a voice command system built using a Raspberry Pi that recognizes disordered speech from people with severe speech disabilities like dysarthria.
2. The system uses Google's speech to text API to convert speech to text which is then processed to match commands. Matched commands trigger modules that provide responses which are converted to speech using a text-to-speech engine.
3. The system was able to successfully take voice commands and trigger the appropriate modules to respond as intended, demonstrating its ability to help those with speech disabilities communicate and interact using voice commands.
Speech Recognition Application for the Speech Impaired using the Android-base...TELKOMNIKA JOURNAL
Those who are speech impaired (tunawicara in the Indonesian language) suffer from
abnormalities in their delivery (articulation) of the language as well their voice in normal speech, resulting
in difficulty in communicating verbally within their environment. Therefore, an application is required that
can help and facilitate conversations for communication. In this research, the authors have developed a
speech recognition application that can recognise speech of the speech impaired, and can translate into
text form with input in the form of sound detected on a smartphone. By using the Google Cloud Speech
Application Programming Interface (API), this allows converting audio to text, and it is also user-friendly to
use such APIs. The Google Cloud Speech API integrates with Google Cloud Storage for data storage.
Although research into speech recognition to text has been widely practiced, this research try to develop
speech recognition, specially for speech impaired's speech, as well as perform a likelihood calculation to
see the factor of tone, pronunciation, and speech speed in speech recognition. The test was conducted by
mentioning the digits 1 through 10. The experimental results showed that the recognition rate for the
speech impaired is about 80%, while the recognition rate for normal speech is 100%.
The project was started with a sole aim in mind that the design should be able to recognize the voice of a person by analyzing the speech signal. The simulation is done in MATLAB. The design of the project is based on using the Linear prediction filter coefficient (LPC) and Principal component analysis (PCA) on data (princomp) for the speech signal analysis. The Sample Collection process is accomplished by using the microphone to record the speech of male/female. After executing the program the speech is analyzed by the analysis part of our MATLAB program code and our design should be able to identify and give the judgment that the recorded speech signal is same as that of our desired output.
Deep Learning for Speech Recognition - Vikrant Singh TomarWithTheBest
Tomar discusses the components of speech recognition, the difference between deep learning for speech and images, system architecture, GMM-HMM based systems, deep neural networks in speech, tandem DNN, and hybrids. There's a lot of exciting stuff to talk about in deep learning communities.
Vikrant Singh Tomar, Founder, Fluent.ai
Speech Recognition Application for the Speech Impaired using the Android-base...TELKOMNIKA JOURNAL
Those who are speech impaired (tunawicara in the Indonesian language) suffer from
abnormalities in their delivery (articulation) of the language as well their voice in normal speech, resulting
in difficulty in communicating verbally within their environment. Therefore, an application is required that
can help and facilitate conversations for communication. In this research, the authors have developed a
speech recognition application that can recognise speech of the speech impaired, and can translate into
text form with input in the form of sound detected on a smartphone. By using the Google Cloud Speech
Application Programming Interface (API), this allows converting audio to text, and it is also user-friendly to
use such APIs. The Google Cloud Speech API integrates with Google Cloud Storage for data storage.
Although research into speech recognition to text has been widely practiced, this research try to develop
speech recognition, specially for speech impaired's speech, as well as perform a likelihood calculation to
see the factor of tone, pronunciation, and speech speed in speech recognition. The test was conducted by
mentioning the digits 1 through 10. The experimental results showed that the recognition rate for the
speech impaired is about 80%, while the recognition rate for normal speech is 100%.
The project was started with a sole aim in mind that the design should be able to recognize the voice of a person by analyzing the speech signal. The simulation is done in MATLAB. The design of the project is based on using the Linear prediction filter coefficient (LPC) and Principal component analysis (PCA) on data (princomp) for the speech signal analysis. The Sample Collection process is accomplished by using the microphone to record the speech of male/female. After executing the program the speech is analyzed by the analysis part of our MATLAB program code and our design should be able to identify and give the judgment that the recorded speech signal is same as that of our desired output.
Deep Learning for Speech Recognition - Vikrant Singh TomarWithTheBest
Tomar discusses the components of speech recognition, the difference between deep learning for speech and images, system architecture, GMM-HMM based systems, deep neural networks in speech, tandem DNN, and hybrids. There's a lot of exciting stuff to talk about in deep learning communities.
Vikrant Singh Tomar, Founder, Fluent.ai
Speech recognition is the next big step that the technology needs to take for general users. An Automatic Speech Recognition (ASR) will play a major role in focusing new technology to users. Applications of ASR are speech to text conversion, voice input in aircraft, data entry, voice user interfaces such as voice dialing. Speech recognition involves extracting features from the input signal and classifying them to classes using pattern matching model. This can be done using feature extraction method. This paper involves a general study of automatic speech recognition and various methods to generate an ASR system. General techniques that can be used to implement an ASR includes artificial neural networks, Hidden Markov model, acoustic –phonetic approach
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using backpropagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using back propagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
Esophageal Speech Recognition using Artificial Neural Network (ANN)Saibur Rahman
Esophageal Speech Recognition using Artificial Neural Network (ANN). In our presentation shows that how to recognize normal speech and Esophageal speech using ANN. We compared our method with other methods and show that our method is better then other method.
Speech to text conversion for visually impaired person using µ law compandingiosrjce
The paper represents the overall design and implementation of DSP based speech recognition and
text conversion system. Speech is usually taken as a preferred mode of operation for human being, This paper
represent voice oriented command for converting into text. We intended to compute the entire speech processing
in real time. This involves simultaneously accepting the input from the user and using software filters to analyse
the data. The comparison was then to be established by using correlation and µ law companding techniques. In
this paper, voice recognition is carried out using MATLAB. The voice command is a person independent. The
voice command is stored in the data base with the help of the function keys. The real time input speech received
is then processed in the speech recognition system where the required feature of the speech words are extracted,
filtered out and matched with the existing sample stored in the database. Then the required MATLAB processes
are done to convert the received data and into text form.
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
Voice input and speech recognition system in tourism/social mediacidroypaes
speech recognition softwares in tourism industry.a computer/mobile analysis of the human voice, especially for the purposes of interpreting words and phrases or identifying an individual voice.
Assistive Examination System for Visually ImpairedEditor IJCATR
This paper presents a design of voice enabled examination system which can be used by the visually challenged students.
The system uses Text-to-Speech (TTS) and Speech-to-Text (STT) technology. The text-to-speech and speech-to-text web based
academic testing software would provide an interaction for blind students to enhance their educational experiences by providing them
with a tool to give the exams. This system will aid the differently-abled to appear for online tests and enable them to come at par with
the other students. This system can also be used by students with learning disabilities or by people who wish to take the examination in
a combined auditory and visual way.
Speech recognition is the next big step that the technology needs to take for general users. An Automatic Speech Recognition (ASR) will play a major role in focusing new technology to users. Applications of ASR are speech to text conversion, voice input in aircraft, data entry, voice user interfaces such as voice dialing. Speech recognition involves extracting features from the input signal and classifying them to classes using pattern matching model. This can be done using feature extraction method. This paper involves a general study of automatic speech recognition and various methods to generate an ASR system. General techniques that can be used to implement an ASR includes artificial neural networks, Hidden Markov model, acoustic –phonetic approach
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using backpropagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using back propagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
Esophageal Speech Recognition using Artificial Neural Network (ANN)Saibur Rahman
Esophageal Speech Recognition using Artificial Neural Network (ANN). In our presentation shows that how to recognize normal speech and Esophageal speech using ANN. We compared our method with other methods and show that our method is better then other method.
Speech to text conversion for visually impaired person using µ law compandingiosrjce
The paper represents the overall design and implementation of DSP based speech recognition and
text conversion system. Speech is usually taken as a preferred mode of operation for human being, This paper
represent voice oriented command for converting into text. We intended to compute the entire speech processing
in real time. This involves simultaneously accepting the input from the user and using software filters to analyse
the data. The comparison was then to be established by using correlation and µ law companding techniques. In
this paper, voice recognition is carried out using MATLAB. The voice command is a person independent. The
voice command is stored in the data base with the help of the function keys. The real time input speech received
is then processed in the speech recognition system where the required feature of the speech words are extracted,
filtered out and matched with the existing sample stored in the database. Then the required MATLAB processes
are done to convert the received data and into text form.
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
Voice input and speech recognition system in tourism/social mediacidroypaes
speech recognition softwares in tourism industry.a computer/mobile analysis of the human voice, especially for the purposes of interpreting words and phrases or identifying an individual voice.
Assistive Examination System for Visually ImpairedEditor IJCATR
This paper presents a design of voice enabled examination system which can be used by the visually challenged students.
The system uses Text-to-Speech (TTS) and Speech-to-Text (STT) technology. The text-to-speech and speech-to-text web based
academic testing software would provide an interaction for blind students to enhance their educational experiences by providing them
with a tool to give the exams. This system will aid the differently-abled to appear for online tests and enable them to come at par with
the other students. This system can also be used by students with learning disabilities or by people who wish to take the examination in
a combined auditory and visual way.
Mini-Project – EECE 365, Spring 2021 You are to read an IlonaThornburg83
Mini-Project – EECE 365, Spring 2021
You are to read an EECE 365-related IEEE article (possibly 2-3 articles if they are on the shorter side), and put together
5-7 slides as if you were to give a 5-8 minute presentation (which you may be asked to give to Amin later in the
semester).
Sample articles (that you may select) have been posted on the course website. As no more than 4 people may select the
same article(s), a sign-up sheet for the sample articles can be found here (to gain access, you may need to first log into
your Wildcat Mail account):
https://docs.google.com/spreadsheets/d/1QTIbVVEUQEGXOMq-brvqAkS5aTDwKccYafpgB49B_ec/edit?usp=sharing
Available articles have been marked in green in the document, and the document will be regularly updated. To
“claim” sample article(s), you MUST email Amin. You are NOT confirmed until you receive a confirmation message
from Amin. Available slots will be filled on a first-come, first-served basis. To minimize back-and-forth emails,
you are encouraged to email Amin with a list of your interested articles, ranked in order of preference.
Note: You are welcome to find other appropriate IEEE article using IEEE Xplore (
https://ieeexplore.ieee.org/Xplore/home.jsp ). Recommended publications include IEEE Signal Processing Magazine,
IEEE Engineering in Medicine and Biology Magazine, IEEE Communications Magazine, or IEEE Spectrum (among
others). If you prefer to pick an article different from the samples posted on the course website, you must first
obtain Amin’s okay for the article.
Use the following questions to guide you through the paper reading process (and subsequently putting together your
slides):
x What are the motivations for the article?
x What is the article’s evaluation of the problem and/or proposed solution?
x What is your analysis of the application of spectrograms to the problem, idea, and/or solution described in the
article?
x What are possible future directions for the topic described in the article?
x What questions are you left with?
The content of your slides should progress/flow in a logical manner, with each point leading to the next in a “smooth”
manner. You should NOT be copying/pasting text from the article to your slides. The text-based content of your slides
should be in your own words, based upon your interpretations/understandings of the article (it is okay if you do not
fully grasp/follow the mathematical equations/proofs in the article). The important thing is for your slides (and
possible presentation to Amin) to convey the “bigger picture” takeaways of the article (in your own words). In addition
to text, you should include appropriate pictures/images in your slides. Pictures that you did not create should be
cited/credited.
This is an “individual” assignment. That is, the content/information contained in one’s slides must be developed/put
together by each individual.
https://docs.google.com/spreadsheets/d ...
Abstract
In this paper we propose a new product inwhich speech is used to interact with computers. Speech is a man’s most powerful form
of communication. The user will be able to give various voice commands to the system, which the system will recognize and
execute tasks based on the input command. This system will provide another form of input (apart from mouse and keyboard) for
daily users. It will also be of great assistance to physically challenged users. The user will be able to perform all the operations
using his voice as input which he is able to perform normally using mouse and keyboard. The system however requires a little bit
training from the user, so that the system will understand the user better. The need of training is due to the fact that every person
has different voice. Also the voice of women is totally distinct from men. More training will result in faster and accurate response.
Extensive experiments are conducted in order to check the accuracy and efficiency of the system, and the results show that our
system is reliable and achieves better accuracy and efficiency than previous systems.
KeyWords: Speech Technology, Voice Response System, Voice User Interface, Voice Recognition.
Advanced Computational Intelligence: An International Journal (ACII)aciijournal
The purpose of this research paper is to illustrate the implementation of a Voice Command System. This
system works on the primary input of a user’s voice. Using voice as an input, we were able to convert it to
text using a speech to text engine. The text hence produced was used for query processing and fetching
relevant information. When the information was fetched, it was then converted to speech using speech to
text conversion and the relevant output to the user was given. Additionally, some extra modules were also
implemented which worked on the concept of keyword matching. These included telling time, weather and
notification from social applications.
VOICE COMMAND SYSTEM USING RASPBERRY PIaciijournal
The purpose of this research paper is to illustrate the implementation of a Voice Command System. This
system works on the primary input of a user’s voice. Using voice as an input, we were able to convert it to
text using a speech to text engine. The text hence produced was used for query processing and fetching
relevant information. When the information was fetched, it was then converted to speech using speech to
text conversion and the relevant output to the user was given. Additionally, some extra modules were also
implemented which worked on the concept of keyword matching. These included telling time, weather and
notification from social applications.
Voice Command System Using Raspberry PIaciijournal
The purpose of this research paper is to illustrate the implementation of a Voice Command System. This
system works on the primary input of a user’s voice. Using voice as an input, we were able to convert it to
text using a speech to text engine. The text hence produced was used for query processing and fetching
relevant information. When the information was fetched, it was then converted to speech using speech to
text conversion and the relevant output to the user was given. Additionally, some extra modules were also
implemented which worked on the concept of keyword matching. These included telling time, weather and
notification from social applications.
In this article, Bhusan Chettri provides an overview of person authentication system that is based on automatic speaker verification technology. He provides background on both the traditional approaches of modelling speakers and current deep learning based approaches. A brief introduction to how these systems can be manipulated is also provided.
Developing a hands-free interface to operate a Computer using voice commandMohammad Liton Hossain
The main focus of this study is to help a handicap person to operate a computer by voice command. It can be used to operate the entire computer functions on the user’s voice commands. It makes use of the Speech Recognition technology that allows the computer system to identify and recognize words spoken by a human using a microphone. This Software will be able to recognize spoken words and enable user to interact with the computer. This interaction includes user giving commands to his computer which will then respond by performing several tasks, actions or operations depending on the commands they gave. For Example: Opening /closing a file in computer, YouTube automation using voice command, Google search using voice command, make a note using voice command, calculation by calculator using voice command etc.
An overview of speaker recognition by Bhusan Chettri.pdfBhusan Chettri
In this article, Bhusan Chettri provides an overview of voice authentication system that is based on automatic speaker verification technology. He provides background on both the traditional approaches of modelling speakers and current deep learning based approaches. A brief introduction to how these systems can be manipulated is also provided.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
The Great Mind Challenge'12
VOICE BASED WEB BROWSER
This document covers the functional and non-functional requirements of the
Voice-Based Web Browser including the physical description of the system as well as the
behavioral and other factors necessary to provide a complete and comprehensive
description of the Voice-Based Web Browser.
Abstract
The technology of voice browsing is rapidly evolving these days. It is because the use of cell phones is increasing at a very high rate, as compared to connected PCs. Listening and speaking are the natural modes of communication and information gathering. As a result we are now heading towards a more voice based approach of browsing rather than operating on textual mode. The command input and the delivery of web contents are entirely in voice. A voice browser is a device: that interprets voice input and interprets voice markup languages to generate voice output. That interprets a script which specifies exactly what to verbally present to the user as well as when to present each piece of information. Benefits Voice is a very natural user interface which speeds up browsing.
Water billing management system project report.pdfKamal Acharya
Our project entitled “Water Billing Management System” aims is to generate Water bill with all the charges and penalty. Manual system that is employed is extremely laborious and quite inadequate. It only makes the process more difficult and hard.
The aim of our project is to develop a system that is meant to partially computerize the work performed in the Water Board like generating monthly Water bill, record of consuming unit of water, store record of the customer and previous unpaid record.
We used HTML/PHP as front end and MYSQL as back end for developing our project. HTML is primarily a visual design environment. We can create a android application by designing the form and that make up the user interface. Adding android application code to the form and the objects such as buttons and text boxes on them and adding any required support code in additional modular.
MySQL is free open source database that facilitates the effective management of the databases by connecting them to the software. It is a stable ,reliable and the powerful solution with the advanced features and advantages which are as follows: Data Security.MySQL is free open source database that facilitates the effective management of the databases by connecting them to the software.
KuberTENes Birthday Bash Guadalajara - K8sGPT first impressionsVictor Morales
K8sGPT is a tool that analyzes and diagnoses Kubernetes clusters. This presentation was used to share the requirements and dependencies to deploy K8sGPT in a local environment.
Understanding Inductive Bias in Machine LearningSUTEJAS
This presentation explores the concept of inductive bias in machine learning. It explains how algorithms come with built-in assumptions and preferences that guide the learning process. You'll learn about the different types of inductive bias and how they can impact the performance and generalizability of machine learning models.
The presentation also covers the positive and negative aspects of inductive bias, along with strategies for mitigating potential drawbacks. We'll explore examples of how bias manifests in algorithms like neural networks and decision trees.
By understanding inductive bias, you can gain valuable insights into how machine learning models work and make informed decisions when building and deploying them.
TOP 10 B TECH COLLEGES IN JAIPUR 2024.pptxnikitacareer3
Looking for the best engineering colleges in Jaipur for 2024?
Check out our list of the top 10 B.Tech colleges to help you make the right choice for your future career!
1) MNIT
2) MANIPAL UNIV
3) LNMIIT
4) NIMS UNIV
5) JECRC
6) VIVEKANANDA GLOBAL UNIV
7) BIT JAIPUR
8) APEX UNIV
9) AMITY UNIV.
10) JNU
TO KNOW MORE ABOUT COLLEGES, FEES AND PLACEMENT, WATCH THE FULL VIDEO GIVEN BELOW ON "TOP 10 B TECH COLLEGES IN JAIPUR"
https://www.youtube.com/watch?v=vSNje0MBh7g
VISIT CAREER MANTRA PORTAL TO KNOW MORE ABOUT COLLEGES/UNIVERSITITES in Jaipur:
https://careermantra.net/colleges/3378/Jaipur/b-tech
Get all the information you need to plan your next steps in your medical career with Career Mantra!
https://careermantra.net/
Hierarchical Digital Twin of a Naval Power SystemKerry Sado
A hierarchical digital twin of a Naval DC power system has been developed and experimentally verified. Similar to other state-of-the-art digital twins, this technology creates a digital replica of the physical system executed in real-time or faster, which can modify hardware controls. However, its advantage stems from distributing computational efforts by utilizing a hierarchical structure composed of lower-level digital twin blocks and a higher-level system digital twin. Each digital twin block is associated with a physical subsystem of the hardware and communicates with a singular system digital twin, which creates a system-level response. By extracting information from each level of the hierarchy, power system controls of the hardware were reconfigured autonomously. This hierarchical digital twin development offers several advantages over other digital twins, particularly in the field of naval power systems. The hierarchical structure allows for greater computational efficiency and scalability while the ability to autonomously reconfigure hardware controls offers increased flexibility and responsiveness. The hierarchical decomposition and models utilized were well aligned with the physical twin, as indicated by the maximum deviations between the developed digital twin hierarchy and the hardware.
Using recycled concrete aggregates (RCA) for pavements is crucial to achieving sustainability. Implementing RCA for new pavement can minimize carbon footprint, conserve natural resources, reduce harmful emissions, and lower life cycle costs. Compared to natural aggregate (NA), RCA pavement has fewer comprehensive studies and sustainability assessments.
ACEP Magazine edition 4th launched on 05.06.2024Rahul
This document provides information about the third edition of the magazine "Sthapatya" published by the Association of Civil Engineers (Practicing) Aurangabad. It includes messages from current and past presidents of ACEP, memories and photos from past ACEP events, information on life time achievement awards given by ACEP, and a technical article on concrete maintenance, repairs and strengthening. The document highlights activities of ACEP and provides a technical educational article for members.
Harnessing WebAssembly for Real-time Stateless Streaming PipelinesChristina Lin
Traditionally, dealing with real-time data pipelines has involved significant overhead, even for straightforward tasks like data transformation or masking. However, in this talk, we’ll venture into the dynamic realm of WebAssembly (WASM) and discover how it can revolutionize the creation of stateless streaming pipelines within a Kafka (Redpanda) broker. These pipelines are adept at managing low-latency, high-data-volume scenarios.
1. International Journal of Scientific Research and Engineering Development-– Volume 2 Issue 1, Mar-Apr 2019
Available at www.ijsred.com
Voice Command System Using Raspberry Pi
M.H. Nandhinee1 Akanksha khan2, S.P.Audline Beena3,Dr.D.Rajinigirinath4
1,2
UG Students, 3
Asst professor, 4
Professor
Department of Computer Science and Engineering
Sri Muthukumaran Institute of Technology,Chennai-69,India
Abstract-Verbal correspondence is an imperative
component in personal satisfaction, anyway upwards of 1.4
percent of people can't use common discourse dependably to
convey their perspectives and sentiments with others which
prompts discourse disadvantages. Discourse handicaps or
discourse hindrances are the parts of correspondence issue in
which the typical discourse get upset like faltering, lips, and so
on. The word incapacity can turn away those individuals who
are experiencing serious discourse inabilities from conveying in
a method for doing thing that enables them to use for one
closures their potential in instruction and entertainment. In
this investigation another type of discourse acknowledgment
framework is produced which perceives the disarranged
discourse of the general population who are experiencing
extreme discourse incapacities. Nowadays, users need a device
or a communication system which is easy to handle. ASR
systems work well for the people who are suffering from severe
speech disabilities such as dysarthria which is the most
common speech disorder and , these studies shows that there is
inverse connection between level of weakness and exactness of
speech recognition. This system describes the development of
speech recognition system which recognizes the disordered
speech.
Keywords—Speech to text, Raspberry Pi, Voice Command
System, Query Processing
I. INTRODUCTION
Talk is the best strategy for social correspondence. Only
5% to 10% of the human people has an absolutelycustomary
strategy for verbal correspondence concerning distinctive
talk features and sound voice; and whatever is leftof the 90
% to 95 % experience the evil impacts of one issue or the
other, for instance, wavering, scattering, dysarthria, apraxia
of talk, etc. Wavering is a talk issue which is affecting a
substantial number of people in their regular day to day
existence. Wavering is an issue that interferes with plain talk.
A person who wavers may reiterate the underlying
fragment of a word (as in wh what) or hold a singular sound
for a long time as in (hhhheeeeellllllooooo). Interventions,
for instance, "um" or "like" can happen likewise; particularly
when they contain repeated ("goodness ohh-ohhho") or
deferred ("ohhhh"). More than 68 million people generally
speaking stammer and has found to impact folks and females
in the extent of 4:1. This issue is depicted by intrusions in the
formation of talk sounds, called disfluencies. This technique
makes it possible to use the speaker's voice and the spoken
word to verify their identity and control access toservices.
II. SYSTEM ARCHITECTURE
A. EXISTING SYSTEM
The current framework experiences the downside that
just predefined voices are conceivable and it can store just
constrained voices. Henceforth, the client can't get the full
data reasonably.
B. PROPOSED SYSTEM
The proposed framework is to such an extent that it can
conquer the downside of the current framework. The venture
configuration include content to discourse. Here whatever
the framework gets as contribution after the order the yield
will get as voice implies discourse.
C. HARDWARE IMPLEMENTATION
Figure(1)
D. MICROPHONE
Amplifier is utilized to take the sound contribution of the
sound. This sound information when further gone through
the framework would be scanned for watchwords. These
watchwords are fundamental for the working of the voice
direction framework as our modules take a shot at the
embodiment of hunting down catchphrases and giving yield
by coordinating catchphrases.
RESEARCH ARTICLE OPEN ACCESS