This document discusses speaker recognition using Mel Frequency Cepstral Coefficients (MFCC). It describes the process of feature extraction using MFCC which involves framing the speech signal, taking the Fourier transform of each frame, warping the frequencies using the mel scale, taking the logs of the powers at each mel frequency, and converting to cepstral coefficients. It then discusses feature matching techniques like vector quantization which clusters reference speaker features to create codebooks for comparison to unknown speakers. The document provides references for further reading on speech and speaker recognition techniques.
Deep Learning techniques have enabled exciting novel applications. Recent advances hold lot of promise for speech based applications that include synthesis and recognition. This slideset is a brief overview that presents a few architectures that are the state of the art in contemporary speech research. These slides are brief because most concepts/details were covered using the blackboard in a classroom setting. These slides are meant to supplement the lecture.
This is a ppt on speech recognition system or automated speech recognition system. I hope that it would be helpful for all the people searching for a presentation on this technology
Deep Learning techniques have enabled exciting novel applications. Recent advances hold lot of promise for speech based applications that include synthesis and recognition. This slideset is a brief overview that presents a few architectures that are the state of the art in contemporary speech research. These slides are brief because most concepts/details were covered using the blackboard in a classroom setting. These slides are meant to supplement the lecture.
This is a ppt on speech recognition system or automated speech recognition system. I hope that it would be helpful for all the people searching for a presentation on this technology
Also known as automatic speech recognition or computer speech recognition which means understanding voice by the computer and performing any required task.
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
This presentation was delivered to a "Web Enabled Business" class at Simon Fraser University in Vancouver. The topic is speech recognition technology, and the presentation covers its origins, how it works, issues, latest trends and future opportunities.
Recently, WaveNet, which predicts the probability distribution of speech sample auto-regressively, provides a new paradigm in speech synthesis tasks.
Since the usage of WaveNet for speech synthesis varies by conditional vectors, it is very important to effectively design a baseline system structure.
In this talk, I would like to first introduce various types of WaveNet vocoders such as conventional speech-domain approach and recently proposed source-filter theory-based approach.
Then, I will explain a linear prediction (LP)-based WaveNet speech synthesis, i.e., LP-WaveNet, which overcomes the limitations of source-filter theory-based WaveNet vocoders caused by the mismatch between speech excitation signal and vocal tract filter.
While presenting experimental setups and results, I also would like to share some know-hows to successfully training the network.
Also known as automatic speech recognition or computer speech recognition which means understanding voice by the computer and performing any required task.
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
This presentation was delivered to a "Web Enabled Business" class at Simon Fraser University in Vancouver. The topic is speech recognition technology, and the presentation covers its origins, how it works, issues, latest trends and future opportunities.
Recently, WaveNet, which predicts the probability distribution of speech sample auto-regressively, provides a new paradigm in speech synthesis tasks.
Since the usage of WaveNet for speech synthesis varies by conditional vectors, it is very important to effectively design a baseline system structure.
In this talk, I would like to first introduce various types of WaveNet vocoders such as conventional speech-domain approach and recently proposed source-filter theory-based approach.
Then, I will explain a linear prediction (LP)-based WaveNet speech synthesis, i.e., LP-WaveNet, which overcomes the limitations of source-filter theory-based WaveNet vocoders caused by the mismatch between speech excitation signal and vocal tract filter.
While presenting experimental setups and results, I also would like to share some know-hows to successfully training the network.
Speaker Recognition System using MFCC and Vector Quantization Approachijsrd.com
This paper presents an approach to speaker recognition using frequency spectral information with Mel frequency for the improvement of speech feature representation in a Vector Quantization codebook based recognition approach. The Mel frequency approach extracts the features of the speech signal to get the training and testing vectors. The VQ Codebook approach uses training vectors to form clusters and recognize accurately with the help of LBG algorithm.
Design and Implementation of Speech Based Scientific CalculatorShantha Suresh M
In recent days speech recognition based systems are gaining more prominence and are used in a wide range of applications. Calculator has become an indispensable part of our lives, used at home, at local retail shops or at workplaces like banks, IT companies, medical laboratories etc. Speech recognized calculator is one of the speech recognition systems, in which the digits and operators spoken by the user with pauses in between are analyzed using various algorithms and finally one of the 10 digits (0 to 9) and one of the 5 operators(+, -, *, /, =) The objective of our paper is to design and implement the speech recognized calculator in MATLAB and Standalone Microcontroller. This designed system is able to take detect the digits and operators of the arithmetic expression whose result is to be calculated. In this paper we have used linear predictive method for recognizing the digit/operator in MATLAB and Analog Speech Recognition Chip.
A Novel, Robust, Hierarchical, Text-Independent Speaker Recognition TechniqueCSCJournals
Automatic speaker recognition system is used to recognize an unknown speaker among several reference speakers by making use of speaker-specific information from their speech. In this paper, we introduce a novel, hierarchical, text-independent speaker recognition. Our baseline speaker recognition system accuracy, built using statistical modeling techniques, gives an accuracy of 81% on the standard MIT database and our baseline gender recognition system gives an accuracy of 93.795%. We then propose and implement a novel state-space pruning technique by performing gender recognition before speaker recognition so as to improve the accuracy/timeliness of our baseline speaker recognition system. Based on the experiments conducted on the MIT database, we demonstrate that our proposed system improves the accuracy over the baseline system by approximately 2%, while reducing the computational time by more than 30%.
Speaker and Speech Recognition for Secured Smart Home ApplicationsRoger Gomes
The paper published in discusses implementation of a robust text-independent speaker recognition system using MFCC extraction of feature vectors its matching using VQ and optimization using LBG, further a text dependent speech recognition system using the DTW algorithm's implementation is discussed in the context of home automation.
Key Trends Shaping the Future of Infrastructure.pdfCheryl Hung
Keynote at DIGIT West Expo, Glasgow on 29 May 2024.
Cheryl Hung, ochery.com
Sr Director, Infrastructure Ecosystem, Arm.
The key trends across hardware, cloud and open-source; exploring how these areas are likely to mature and develop over the short and long-term, and then considering how organisations can position themselves to adapt and thrive.
Builder.ai Founder Sachin Dev Duggal's Strategic Approach to Create an Innova...Ramesh Iyer
In today's fast-changing business world, Companies that adapt and embrace new ideas often need help to keep up with the competition. However, fostering a culture of innovation takes much work. It takes vision, leadership and willingness to take risks in the right proportion. Sachin Dev Duggal, co-founder of Builder.ai, has perfected the art of this balance, creating a company culture where creativity and growth are nurtured at each stage.
Software Delivery At the Speed of AI: Inflectra Invests In AI-Powered QualityInflectra
In this insightful webinar, Inflectra explores how artificial intelligence (AI) is transforming software development and testing. Discover how AI-powered tools are revolutionizing every stage of the software development lifecycle (SDLC), from design and prototyping to testing, deployment, and monitoring.
Learn about:
• The Future of Testing: How AI is shifting testing towards verification, analysis, and higher-level skills, while reducing repetitive tasks.
• Test Automation: How AI-powered test case generation, optimization, and self-healing tests are making testing more efficient and effective.
• Visual Testing: Explore the emerging capabilities of AI in visual testing and how it's set to revolutionize UI verification.
• Inflectra's AI Solutions: See demonstrations of Inflectra's cutting-edge AI tools like the ChatGPT plugin and Azure Open AI platform, designed to streamline your testing process.
Whether you're a developer, tester, or QA professional, this webinar will give you valuable insights into how AI is shaping the future of software delivery.
Dev Dives: Train smarter, not harder – active learning and UiPath LLMs for do...UiPathCommunity
💥 Speed, accuracy, and scaling – discover the superpowers of GenAI in action with UiPath Document Understanding and Communications Mining™:
See how to accelerate model training and optimize model performance with active learning
Learn about the latest enhancements to out-of-the-box document processing – with little to no training required
Get an exclusive demo of the new family of UiPath LLMs – GenAI models specialized for processing different types of documents and messages
This is a hands-on session specifically designed for automation developers and AI enthusiasts seeking to enhance their knowledge in leveraging the latest intelligent document processing capabilities offered by UiPath.
Speakers:
👨🏫 Andras Palfi, Senior Product Manager, UiPath
👩🏫 Lenka Dulovicova, Product Program Manager, UiPath
PHP Frameworks: I want to break free (IPC Berlin 2024)Ralf Eggert
In this presentation, we examine the challenges and limitations of relying too heavily on PHP frameworks in web development. We discuss the history of PHP and its frameworks to understand how this dependence has evolved. The focus will be on providing concrete tips and strategies to reduce reliance on these frameworks, based on real-world examples and practical considerations. The goal is to equip developers with the skills and knowledge to create more flexible and future-proof web applications. We'll explore the importance of maintaining autonomy in a rapidly changing tech landscape and how to make informed decisions in PHP development.
This talk is aimed at encouraging a more independent approach to using PHP frameworks, moving towards a more flexible and future-proof approach to PHP development.
Accelerate your Kubernetes clusters with Varnish CachingThijs Feryn
A presentation about the usage and availability of Varnish on Kubernetes. This talk explores the capabilities of Varnish caching and shows how to use the Varnish Helm chart to deploy it to Kubernetes.
This presentation was delivered at K8SUG Singapore. See https://feryn.eu/presentations/accelerate-your-kubernetes-clusters-with-varnish-caching-k8sug-singapore-28-2024 for more details.
LF Energy Webinar: Electrical Grid Modelling and Simulation Through PowSyBl -...DanBrown980551
Do you want to learn how to model and simulate an electrical network from scratch in under an hour?
Then welcome to this PowSyBl workshop, hosted by Rte, the French Transmission System Operator (TSO)!
During the webinar, you will discover the PowSyBl ecosystem as well as handle and study an electrical network through an interactive Python notebook.
PowSyBl is an open source project hosted by LF Energy, which offers a comprehensive set of features for electrical grid modelling and simulation. Among other advanced features, PowSyBl provides:
- A fully editable and extendable library for grid component modelling;
- Visualization tools to display your network;
- Grid simulation tools, such as power flows, security analyses (with or without remedial actions) and sensitivity analyses;
The framework is mostly written in Java, with a Python binding so that Python developers can access PowSyBl functionalities as well.
What you will learn during the webinar:
- For beginners: discover PowSyBl's functionalities through a quick general presentation and the notebook, without needing any expert coding skills;
- For advanced developers: master the skills to efficiently apply PowSyBl functionalities to your real-world scenarios.
Transcript: Selling digital books in 2024: Insights from industry leaders - T...BookNet Canada
The publishing industry has been selling digital audiobooks and ebooks for over a decade and has found its groove. What’s changed? What has stayed the same? Where do we go from here? Join a group of leading sales peers from across the industry for a conversation about the lessons learned since the popularization of digital books, best practices, digital book supply chain management, and more.
Link to video recording: https://bnctechforum.ca/sessions/selling-digital-books-in-2024-insights-from-industry-leaders/
Presented by BookNet Canada on May 28, 2024, with support from the Department of Canadian Heritage.
Search and Society: Reimagining Information Access for Radical FuturesBhaskar Mitra
The field of Information retrieval (IR) is currently undergoing a transformative shift, at least partly due to the emerging applications of generative AI to information access. In this talk, we will deliberate on the sociotechnical implications of generative AI for information access. We will argue that there is both a critical necessity and an exciting opportunity for the IR community to re-center our research agendas on societal needs while dismantling the artificial separation between the work on fairness, accountability, transparency, and ethics in IR and the rest of IR research. Instead of adopting a reactionary strategy of trying to mitigate potential social harms from emerging technologies, the community should aim to proactively set the research agenda for the kinds of systems we should build inspired by diverse explicitly stated sociotechnical imaginaries. The sociotechnical imaginaries that underpin the design and development of information access technologies needs to be explicitly articulated, and we need to develop theories of change in context of these diverse perspectives. Our guiding future imaginaries must be informed by other academic fields, such as democratic theory and critical theory, and should be co-developed with social science scholars, legal scholars, civil rights and social justice activists, and artists, among others.
The Art of the Pitch: WordPress Relationships and SalesLaura Byrne
Clients don’t know what they don’t know. What web solutions are right for them? How does WordPress come into the picture? How do you make sure you understand scope and timeline? What do you do if sometime changes?
All these questions and more will be explored as we talk about matching clients’ needs with what your agency offers without pulling teeth or pulling your hair out. Practical tips, and strategies for successful relationship building that leads to closing the deal.
2. HUMAN SPEECH
•
The human speech contains numerous
discriminative features that can be used to
identify speakers.
•
Speech contains significant energy from
zero frequency up to around 5 kHz.
•
Objective of automatic speaker recognition
is to extract, characterize and recognize
the information about speaker identity.
•
The property of speech signal changes
markedly as a function of time.
3. SPEECH DISCERNMENT
•
Speaker recognition systems
contain two main modules:
•
•
•
feature extraction
feature matching
Feature extraction:
Extract a small amount of data
from the voice signal that can
be used to represent each
speaker
•
Feature matching:
Procedure to identify the
unknown
speaker
by
comparing extracted features
from his/her voice input with
the ones from a set of known
speakers
5. INTRODUCTION
•
Speech signal - slowly timed
varying signal (it is called quasistationary)
•
Signal-processing front end
Conversion of speech
waveform, using digital signal
processing (DSP) tools, to a set of
features (at a considerably lower
information rate) for further
analysis
•
Short-time spectral analysis
Characterization of the speech
signal
6. CHOICE OF METHODOLOGY
•
Linear Predictive Coding (LPC)
•
Mel-Frequency Cepstrum
Coefficients (MFCC)
•
Perceptual Linear Predictive
Analysis (PLP)
8. MEL FREQUENCY CEPSTRUM CO-EFFICIENT (MFCC)
Main purpose of the MFCC processor is to mimic the behavior of the human ears
MFFC‟s - less susceptible to variations
Speech input typically recorded at a sampling rate above 10000 Hz
This sampling frequency chosen to minimize the effects of aliasing in the analog-to-digital
conversion
• Sampled signals can capture all frequencies up to 5 kHz, which cover most energy of
sounds that are generated by humans
•
•
•
•
9. FRAME BLOCKING
•
In this step the continuous speech signal is
blocked into frames of N samples, with
adjacent frames being separated by M (M <
N)
•
The first frame consists of the first N samples
•
The second frame begins M samples after
the first frame, and overlaps it by N - M
samples
•
Process continues until all the speech is
accounted for within one or more frames
•
Taken values for N and M are N = 256 and M
= 100
•
The result after this step is referred to as
spectrum or periodogram.
10. MEL-FREQUENCY WRAPPING
•
For each tone with an actual frequency, f, a
subjective pitch is measured on a scale called the
„mel‟ scale.
•
Mel-frequency scale is a linear frequency spacing
below 1000 Hz and a logarithmic spacing above
1000 Hz
•
One approach to simulating the subjective
spectrum is to use a filter bank, spaced uniformly
on the mel-scale
11. MEL-FREQUENCY WRAPPING
•
That filter bank has a triangular bandpass
frequency response, and the spacing as well as
the bandwidth is determined by a constant mel
frequency interval
•
The number of mel spectrum coefficients, K, is
typically chosen as 20.
•
Filter bank is applied in the frequency domain
12. CEPSTRUM
•
In this final step, we convert the log mel spectrum back to time
•
Result - mel frequency cepstrum coefficients (MFCC)
•
Cepstral representation of speech spectrum provides good representation of the local
spectral properties of the signal for the given frame analysis
•
Mel spectrum coefficients (and so their logarithm) are real numbers
can be converted to the time domain using the Discrete Cosine Transform (DCT)
•
We exclude the first component, from the DCT since it represents the mean value of the
input signal, which carries little speaker specific information
14. FEATURE MATCHING
•
Comes under pattern recognition (The objects of interest are generically called patterns)
•
Patterns - sequences of acoustic vectors that are extracted from an input speech using
extraction
•
Test Set - Patterns used to test the classification algorithm
•
Feature matching techniques used in speaker recognition - Dynamic Time Warping
(DTW), Hidden Markov Modeling (HMM), and Vector Quantization (VQ)
•
VQ approach used due to:
• ease of implementation
• high accuracy
VQ is a process of mapping vectors from a large vector space to a finite number of
regions in that space.
•
•
Each region is called a cluster and can be represented by its center called a codeword.
The collection of all codewords is called a codebook.
15. VECTOR QUANTIZATION CODE-BOOK FORMATION
•
•
•
•
Two speakers and two dimensions
of the acoustic space
Circles - acoustic vectors from the
speaker 1
Triangles - acoustic vectors from
speaker 2
Training phase
• using the clustering algorithm
a
speaker-specific
VQ
codebook is generated for
each known speaker by
clustering his/her training
acoustic vectors
• Result codewords (centroids) black circles and black
triangles for speaker 1 and 2
16. VECTOR QUANTIZATION CODE-BOOK FORMATION
•
•
•
Distance from a vector to the
closest codeword of a codebook is
called VQ-distortion
Input utterance of an unknown
voice is “vector-quantized” using
each trained codebook and the
total VQ distortion is computed
The speaker corresponding to the
VQ codebook with smallest total
distortion is identified as the
speaker of the input utterance.
19. L.R. Rabiner and B.H. Juang, Fundamentals of Speech
Recognition, Prentice-Hall, Englewood Cliffs, N.J., 1993.
L.R Rabiner and R.W. Schafer, Digital Processing of Speech
Signals, Prentice-Hall, Englewood Cliffs, N.J., 1978.
REFERENCES
S.B. Davis and P. Mermelstein, “Comparison of parametric
representations for monosyllabic word recognition in continuously spoken
sentences”, IEEE Transactions on Acoustics, Speech, Signal
Processing, Vol. ASSP-28, No. 4, August 1980.
Y. Linde, A. Buzo & R. Gray, “An algorithm for vector quantizer
design”, IEEE Transactions on Communications, Vol. 28, pp.84-95, 1980.
S. Furui, “Speaker independent isolated word recognition using
dynamic features of speech spectrum”, IEEE Transactions on
Acoustic, Speech, Signal Processing, Vol. ASSP-34, No. 1, pp. 5259, February 1986.
S. Furui, “An overview of speaker recognition technology”, ESCA
Workshop on Automatic Speaker Recognition, Identification and
Verification, pp. 1-9, 1994.
F.K. Song, A.E. Rosenberg and B.H. Juang, “A vector
quantisation approach to speaker recognition”, AT&T Technical
Journal, Vol. 66-2, pp. 14-26, March 1987.
comp.speech Frequently Asked Questions WWW site,
http://svr-www.eng.cam.ac.uk/comp.speech/
Editor's Notes
An example of speech signal is shown in Figure 2. When examined over a sufficiently short period of time (between 5 and 100 msec), its characteristics are fairly stationary. However, over long periods of time (on the order of 1/5 seconds or more) the signal characteristic change to reflect the different speech sounds being spoken. Therefore, short-time spectral analysis is the most common way to characterize the speech signal.
An example of speech signal is shown in Figure 2. When examined over a sufficiently short period of time (between 5 and 100 msec), its characteristics are fairly stationary. However, over long periods of time (on the order of 1/5 seconds or more) the signal characteristic change to reflect the different speech sounds being spoken. Therefore, short-time spectral analysis is the most common way to characterize the speech signal.
MFCC’s are based on the known variation of the human ear’s critical bandwidths with frequency, filters spaced linearly at low frequencies and logarithmically at high frequencies have been used to capture the phonetically important characteristics of speech. This is expressed in the mel-frequency scale, which is a linear frequency spacing below 1000 Hz and a logarithmic spacing above 1000 Hz. The process of computing MFCCs is described in more detail next.
As mentioned above, psychophysical studies have shown that human perception of the frequency contents of sounds for speech signals does not follow a linear scale. A useful way of thinking about this mel-wrapping filter bank is to view each filter as a histogram bin (where bins have overlap) in the frequency domain.
As mentioned above, psychophysical studies have shown that human perception of the frequency contents of sounds for speech signals does not follow a linear scale. A useful way of thinking about this mel-wrapping filter bank is to view each filter as a histogram bin (where bins have overlap) in the frequency domain.