Report Sample
Report Sample
CHAPTER 1
INTRODUCTION
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CHAPTER 2
LITERATURE SURVEY
This paper has presented a new view of a synthesis database for use in unit
concatenative speech synthesis.The units in a synthesis database can be treated as
states in a state transition network with the state occupancy costs given by the
target cost, and the state transition costs given by the concenatanation of pairs of
units. Two methods have been presented for training the target and concatenation
costs, weight space search and regrewtion training. The regrewtion training
method is more effective because of its substantially lower computational
requirements and greater flexibility.
2.2 Recognition of noisy speech using Dynamic spectral sub band centroids
They introduced two novel types of linguistic features for training the
multilingual parametric acoustic models for text-to-speech synthesis: areal and
phylogenetic features. Although intuitively, such features should have a positive
contribution to the overall synthesis quality, we showed that such claim is at
present inconclusive. Out of diverse set of nine languages we were able to
positively confirm this hypothesis for one language only ( Romanion ).
This project aims to provide an easy platform to learn and master the
English language with modern ways of technology. It includes the correctness of
spelling and meaning with end results of achieving excellence in pronunciation.
In future we are planning to improve the pronunciation i.e. sound accuracy by
incorporating appropriate filtering techniques. Comparative study of the existing
TTS and STT algorithms are performed and work has to be done to improve the
performance & improve the quality of the output. The project has been planned
to be desgined in a way that it is the complete course learning process for the
betterment of pronunciation of the users struggling to achieve at convenience.
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2.5 Language Translator Application
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CHAPTER 3
BLOCK DIAGRAM
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The proposed system Universal Speech Interface integrates three key
functionalities: Text-to-Speech (TTS), Speech-to-Text (STT) conversion,
and translation. Here's how the system would typically work:
2. Main Execution (if _name_ == "_main_":): The script prompts the user
to choose between performing speech recognition and translation or text-
to-speech conversion.
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CHAPTER 4
EXISTING SYSTEM
The existing system may work in foreign slang which may not be easily
understandable for the users. The Predefined slang used is used in the system,
which cannot be changed. This causes need help from others or to hear it many
times to understand. The existing system may only works in a single language
and cannot translate into every Indian native languages. The existing system
may not function all the three functions in a same device: Text-To-Speech
convertion, Voice-To-Text convertion and translation. This makes it harder for
the users like blind people.
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CHAPTER 5
PROPOSED SYSTEM
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text-to-speech conversion, each serving a pivotal role in enhancing
communication accessibility and efficiency. The speech recognition module,
powered by the speech_recognition library, captures spoken words from the
microphone, transcribing them into text format. This functionality enables users
to dictate text, issue voice commands, or transcribe spoken content. The
translation module, utilizing the googletrans library, facilitates seamless
communication across language barriers by translating text from one language to
another. This feature empowers users to engage in multilingual conversations and
comprehend content in various languages. Finally, the text-to-speech module,
leveraging the pyttsx3 library, transforms written text into natural-sounding
speech, enabling users to listen to translated text or synthesized content. Together,
these modules provide a versatile platform for inclusive communication,
accommodating diverse linguistic preferences and accessibility needs.
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CHAPTER 6
REQUIREMENTS
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CHAPTER 7
SOFTWARE DESCRIPTION
7.1 PYTHON
Fig.7.1 Python
High-Level:
Interpreted:
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Simplicity and Readability:
Web Development:
Frameworks like Django and Flask are popular for building web
applications and APIs.
Libraries like NumPy, pandas, and Matplotlib make it easy to work with
and visualize data.
Libraries like TensorFlow, PyTorch, and scikit-learn are widely used for
building machine learning models and AI applications.
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7.1.1 FEATURES:
Python is syntax is designed to be simple and easy to read, with a clear and
consistent structure. This makes it particularly suitable for beginners and
experienced programmers alike.
Dynamic Typing:
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Cross-Platform Compatibility:
Memory Management:
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7.2 SPEECH RECOGNITION:
Multi-engine Support:
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Audio Input Sources:
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• Language Support: The library supports recognition in multiple
languages and allows users to specify the language of the input speech.
This enables recognition of speech in different languages and accents.
• Customization: Users can customize recognition parameters such as
language models, recognition confidence thresholds, and audio input
sources to optimize performance for specific use cases and environments.
Java developers can leverage the Java Speech API (JSAPI), along with
libraries like Sphinx4 and CMU PocketSphinx, for robust speech recognition in
Java applications. In the C# ecosystem, Microsoft's System. Speech namespace
in the .NET Framework and third-party libraries such as NAudio and Microsoft
Speech Platform SDK offer comprehensive speech recognition solutions.
C++ developers can utilize the Microsoft Speech API (SAPI) or bindings
for open-source engines like PocketSphinx and CMU Sphinx for speech
recognition tasks. For iOS development in Swift, the Speech framework provides
native support, while Android developers can rely on the Android
SpeechRecognizer class in Kotlin applications.
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Community and Documentation:
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Audio Input Sources:
Adjustable Parameters:
Cross-Platform Compatibility:
Efficient performance:
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refinement, resulting in accurate recognition with minimal computational
burden.
Scalability:
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Comprehensive documentation, tutorials, and guides further support
developers in understanding and using the library effectively. Regular updates
and releases ensure that the library stays current with the latest advancements in
speech recognition technology. Overall, the active community and robust
support resources contribute to the success and continued improvement of the
speech recognition library.
Voice-Controlled Applications:
Transcription Services:
Language Translation:
Voice Search:
Voice search applications like Google Voice Search and Apple's Siri
Search allow users to search the internet using spoken queries. Speech
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recognition technology interprets the user's voice commands, converts them into
text, and retrieves relevant search results, streamlining information retrieval and
enhancing user experience
Dictation Software:
Accessibility Tools:
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Automated Captioning:
Voice-Controlled Gaming:
Automotive Interfaces:
Healthcare:
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quickly and accurately transcribe medical notes, reducing administrative
burden and improving documentation accuracy.
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fig.7.5 Speech recognition algorithm code
Signal Preprocessing:
The speech signal is preprocessed to remove noise and enhance its quality.
Techniques such as noise reduction, filtering, and normalization may be applied
to improve signal-to-noise ratio.
Feature Extraction:
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Acoustic Modeling:
Language Modeling:
Decoding:
The acoustic and language models are combined to decode the sequence of
feature vectors into a sequence of words or sentences. Techniques such as
Dynamic Time Warping (DTW), Viterbi decoding, or beam search may be used
for decoding.
Post-processing:
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functionality into their projects, facilitating cross-lingual communication and
content localization.
Text Detection:
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Translation Accuracy:
Ease of Use:
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Bindings for Multiple Programming Languages:
It's important to note that when directly interacting with the Google
Translate API, developers need to adhere to the API usage limits, authentication
requirements, and terms of service set forth by Google. Additionally, the
availability and stability of the Google Translate API may vary over time, so it's
essential to stay updated on any changes or deprecations to the API.
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Community and Documentation:
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Chatbots and Virtual Assistants:
E-commerce Platforms:
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Social Media and Communication Platforms:
Entertainment Industry:
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applications. Its rich features, ease of use, and broad applicability make it a
valuable asset in projects requiring multilingual support and cross-lingual
communication.
The googletrans library itself does not implement any specific algorithms
for translation. Instead, it serves as a Python wrapper for Google's Translation
API, which is a sophisticated system built on various algorithms and techniques
for language translation. Here's an overview of the general process and
algorithms involved in the translation process, as facilitated by the Google
Translate API
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Fig.7.8 Googletrans Library Algorithm Code
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Language Detection:
Translation Models:
Website Localization:
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Cross-Lingual Data Analysis:
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7.4 PYTTSX3
The library provides a simple and intuitive API for controlling speech
synthesis, allowing developers to customize speech parameters such as voice,
rate, and volume.
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7.4.1 FEATURES OF PYTTSX3 :
Cross-Platform Compatibility:
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applications where auditory feedback enhances user experience, such as virtual
assistants, accessibility tools, and educational software.
Pyttsx3 offers a simple and intuitive API for controlling speech synthesis
operations. The API is designed to be easy to use, allowing developers to quickly
get started with incorporating TTS functionality into their projects. With
straightforward methods and parameters, developers can initiate speech
synthesis, adjust speech parameters, and handle events during the synthesis
process with minimal effort.
Event Handling:
Concurrency Support:
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Integration with NumPy:
Flexible Configuration:
Accessibility:
Educational Software:
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language learning, literacy development, and accessibility in educational
environments.
Accessibility Tools:
Educational Software:
Virtual Assistants:
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hands-free interaction and voice-controlled functionalities. Virtual assistants
powered by Pyttsx3 can perform a wide range of tasks, including providing
information, setting reminders, managing schedules, and controlling smart home
devices.
Multimedia Applications:
Voice-Controlled Interfaces:
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7.4.3 PYTTSX3 ALGORITHM:
Initialization:
Text Input:
With the TTS engine initialized, developers provide the text they want to
convert into speech. This text can be dynamically generated or retrieved from
external sources like files or user input.
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Speech Synthesis:
Once the text is provided, developers use the say() method of the TTS
engine instance to convert the text into audible speech. This method accepts the
input text as a parameter and triggers the TTS engine to synthesize the speech.
pyttsx3 supports multiple TTS engines, each with its own set of features
and capabilities. The library automatically selects the appropriate TTS engine
based on the platform and system configuration. For example, on Windows, it
may use the SAPI5 engine, while on macOS, it may use NSSpeechSynthesizer.
Speech Generation:
The selected TTS engine processes the input text and generates audio
output representing the spoken words. This process involves converting the text
into phonetic representations, applying intonation and prosody rules, and
synthesizing the speech waveform.
Audio Playback:
Once the speech is synthesized, pyttsx3 plays back the audio through the
system's audio output device, such as speakers or headphones. Developers can
control various aspects of the speech output, such as volume, rate (speed), pitch,
and voice selection, using the methods provided by the pyttsx3 library.
Asynchronous Operation:
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Customization:
Event Handling:
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7.5 PyAudio:
Cross-Platform Compatibility:
Flexible Configuration:
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through various channels. PyAudio's GitHub repository serves as the primary
hub for community interaction, bug reporting, feature requests, and code
contributions.
Here, developers can browse through issues, submit bug reports, and
contribute code improvements via pull requests. Additionally, Stack Overflow
provides a popular platform for asking questions, seeking help, and sharing
knowledge related to PyAudio usage.
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commands, perform natural language processing, and generate synthesized
speech responses.
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Music Production and Editing:
Gaming Industry:
Accessibility Tools:
To support audio input from the microphone, you can install the "pyaudio"
library using the command "pip install pyaudio". On macOS, if issues arise
during installation, it might be necessary to install Homebrew first and then use
it to install "portaudio", a dependency of "pyaudio".
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Step 4: Installing the googletrans 4.0.0-rc1 library
To install googletrans version 4.0.0-rc1, you can use pip, the Python
package manager. Open your command-line interface and execute the following
command:
To install the pyttsx3 library, you can use pip, which is the Python package
manager. First, open the command-line interface on your operating system. Then,
execute the command "pip install pyttsx3".
This will automatically download and install the pyttsx3 library along with
any necessary dependencies from the Python Package Index (PyPI). Once the
installation is complete, you can verify that pyttsx3 is installed correctly by
running a simple test script. This script initializes the pyttsx3 engine, speaks a
predefined text using the engine, and waits for the speech to finish. If you hear
the spoken text, then pyttsx3 is installed and functioning properly. You can now
utilize pyttsx3 in your Python scripts to convert text into speech effortlessly.
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Below is the implementation:
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CHAPTER 8
RESULT
INPUT
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OUTPUT
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CHAPTER 9
CONCLUSION
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FUTURE ENHANCEMENT:
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REFERENCES
[1] Hunt A.J. and BlackA.W., “Unit selectionin a concatenative speech synthesis
system for a large speech database,” in Proceedings of IEEE Int. Conf. Acoust.,
Speech, and Signal Processing, 1996, pp. 373–376.
[2] Jose D V, Alfateh Mustafa, Sharan R, "A Novel Model for Speech to Text
Conversion," International Refereed Journal of Engineering and Science
(IRJES), vol 3, no. 1, 2014.
[4] Sproat R., Black A.W., Chen S., Kumar S., Ostendorf M., and Richards C.,
“Normalizationof non-standardwords,” Computer Speech and Language, pp.
287–333, 2001.
[6] Black A.W., Zen H., and Tokuda K., “Statistical parametric speech synthesis,”
in Proceedings of IEEE Int. Conf. Acoust., Speech, and Signal Processing,
Honolulu, USA, 2007.
[8] Sireesh Haang Limbu, “Direct Speech to Speech Translation Using Machine
Learning”, December 2020
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[9] Kunal Sachdev, Hrishabh Srivastava, Sambhav Jain, Dipti Mishra
Sharma "Hindi to English Machine Translation: Using Effective Selection in
Multi-model SMT " LREC 2014.
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