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Unit_2_NMU

The project aims to build a comprehensive speech recognition pipeline that incorporates noise reduction, feature extraction, and acoustic modeling using advanced techniques like HMMs and deep learning. It targets various domains such as healthcare and customer service, with applications including call center automation and accessibility tools. The project will culminate in a fully functional model, performance evaluation metrics, and a Power BI dashboard for stakeholders, all to be completed within 10 days.

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balamurugan532005
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15 views4 pages

Unit_2_NMU

The project aims to build a comprehensive speech recognition pipeline that incorporates noise reduction, feature extraction, and acoustic modeling using advanced techniques like HMMs and deep learning. It targets various domains such as healthcare and customer service, with applications including call center automation and accessibility tools. The project will culminate in a fully functional model, performance evaluation metrics, and a Power BI dashboard for stakeholders, all to be completed within 10 days.

Uploaded by

balamurugan532005
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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Project Title Building an End-to-End Speech Recognition

Pipeline: Signal Processing, Acoustic Modeling,


and Performance Evaluation

Skills take away From This Project Noise Reduction Techniques, Feature
Extraction, Machine Learning and Deep
Learning Skills, Data Preprocessing and
Analysis Skills, Signal Processing

Domain Healthcare, Customer Service, Accessibility


Tools, IoT and Smart Devices, Education and
E-Learning, Entertainment and Media,
Automotive, Security and Surveillance, Retail
and E-Commerce, Telecommunications

Problem Statement:

Speech recognition systems are critical for applications like virtual assistants,
transcription services, and voice-controlled devices. However, raw audio signals
often contain background noise, making accurate speech recognition
challenging. Additionally, extracting meaningful features from audio signals and
building robust acoustic models require advanced signal processing and
machine learning techniques.

The goal of this project is to design and implement a complete speech


recognition pipeline that includes noise reduction, feature extraction (e.g.,
MFCCs), voice activity detection (VAD), and acoustic modeling using Hidden
Markov Models (HMMs) and deep learning techniques. The system will be
evaluated for accuracy and performance.

Business Use Cases:


1.​ Call Center Automation
a.​ Automate transcription and sentiment analysis of customer calls.
2.​ Accessibility Tools
a.​ Develop tools for individuals with hearing impairments by
converting spoken content into readable text.
3.​ Voice Assistants
a.​ Enhance the accuracy of voice assistants in understanding user
commands across different accents and environments.
4.​ Meeting Transcription
a.​ Provide real-time transcription services for business meetings,
enabling better record-keeping and collaboration.
5.​ Voice-Controlled Devices
a.​ Enhance the reliability of voice commands in IoT devices.
Approach:

Data Collection and Cleaning

●​ Collect a speech corpus dataset containing clean and noisy audio


samples.
●​ Preprocess the data by normalizing volume levels, removing silence,
and segmenting audio into frames.
●​ Apply noise reduction techniques (e.g., spectral subtraction, Wiener
filtering).

Data Analysis

●​ Extract features such as MFCCs, pitch, and energy from the


preprocessed audio signals.
●​ Perform Voice Activity Detection (VAD) to identify speech segments
and discard non-speech portions.

Visualization

●​ Visualize spectrograms of raw and processed audio signals.


●​ Plot MFCCs and other extracted features to understand their
distribution.
●​ Compare noise-reduced signals with original signals using waveforms.

Advanced Analytics

●​ Train a Hidden Markov Model (HMM) for acoustic modeling using the
extracted features.
●​ Implement a simple deep learning model (e.g., CNN or RNN) for
comparison.
●​ Evaluate the performance of both models using metrics like Word
Error Rate (WER) and accuracy.

Power BI Integration

Use Power BI to create dashboards showing:

●​ Accuracy metrics of different models.


●​ Comparison of noise reduction techniques.
●​ Feature distributions and correlations

Visualization
●​ Waveform Plots : Raw vs. noise-reduced audio signals.
●​ Spectrograms : Time-frequency representation of audio.
●​ Feature Plots : MFCCs, pitch, and energy distributions.
●​ Accuracy Metrics : Bar charts comparing HMM and deep learning model
performance.
●​ Power BI Dashboard : Interactive visualizations for business stakeholders.

Exploratory Data Analysis (EDA)


●​ Analyze the distribution of audio durations and sampling rates.
●​ Identify common types of noise in the dataset.
●​ Explore the correlation between extracted features (e.g., MFCCs and
pitch).
●​ Evaluate the effectiveness of VAD in isolating speech segments.
●​ Compare the performance of different noise reduction techniques.

Results

The results should include:


●​ A speech recognition pipeline that effectively reduces noise and extracts
meaningful features.
●​ An acoustic model trained using HMMs and deep learning techniques.
●​ Improved accuracy compared to baseline models.
●​ Insights into the strengths and weaknesses of traditional vs. modern
approaches.
Recommendation to End User

●​ For real-time applications, use deep learning-based models due to their


superior accuracy.
●​ For resource-constrained environments, HMMs can provide a lightweight
alternative.
●​ Continuously update the model with new data to improve generalization.

Project Evaluation

●​ Word Error Rate (WER): Calculate the percentage of incorrectly predicted


words. Formula: WER = (Substitutions+Deletions+Insertions ) / Total Words
●​ Accuracy : Percentage of correctly recognized words.
●​ Precision, Recall, F1-Score : Evaluate the performance of VAD.
●​ Signal-to-Noise Ratio (SNR) : Assess the effectiveness of noise
reduction techniques.
●​ Training Time and Inference Latency : Measure the efficiency of
the models.

Data Set:
Data Set Link: Data
Data Set Explanation:
●​ A large-scale corpus of read English speech derived from audiobooks.
●​ Audio is sampled at 16 kHz, ensuring high-quality recordings.
●​ It is split into clean and noisy subsets for varied conditions.
●​ Subsets include 100-hour, 360-hour, and 500-hour splits for scalability.
●​ Transcriptions are manually curated and aligned with audio clips.
●​ Metadata includes speaker IDs and chapter information for additional
tasks.
●​ Preprocessed train-test splits facilitate easy benchmarking of ASR
models.
●​ Supports research in speaker verification, language modeling, and
synthesis.
●​ Metadata including speaker information and chapter details.
●​ Usage : Ideal for training and evaluating acoustic models.
Project Deliverables:

●​ Source Code
●​ A trained speech-to-text transcription model.
●​ A Power BI dashboard showcasing performance metrics.
●​ A report summarizing EDA findings, model performance, and evaluation
metrics.
●​ A fully functional speech recognition pipeline.
●​ A report detailing the methodology, results, and recommendations.
●​ A Power BI dashboard for business stakeholders.
●​ Pipeline Creation for the streamless execution of the problem statement

Timeline:

The project must be completed and submitted within 10 days from the assigned
date.

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