An Evolutionary Approach to Speech Quality Estimationadil raja
This document presents research using genetic programming to develop non-intrusive models for estimating voice over IP (VoIP) quality. Researchers used a VoIP simulation environment to generate distorted speech files under different network conditions and trained genetic programs to map transport layer metrics like packet loss and delay to mean opinion scores. The best models achieved good accuracy compared to the intrusive PESQ standard with only 1-3 variables, making them suitable for real-time VoIP quality monitoring. Future work aims to include wideband codecs and develop a unified quality estimation model.
The document discusses methods for objective and subjective video quality assessment and speech enhancement. It covers four parts: (1) a classification and review of no-reference visual quality assessment methods, (2) no-reference and reduced-reference methods for video quality assessment including neural network and support vector machine approaches, (3) subjective methods for video quality assessment including studies on low resolution videos and crowdsourcing, and (4) speech enhancement techniques including spectral center-of-gravity based demodulation and convex optimization based demodulation. The document evaluates various computational models and machine learning techniques for video and speech quality assessment.
05 comparative study of voice print based acoustic features mfcc and lpccIJAEMSJORNAL
Voice is the best biometric feature for investigation and authentication. It has both biological and behavioural features. The acoustic features are related to the voice. The Speaker Recognition System is designed for the automatic authentication of speaker’s identity which is truly based on the human’s voice. Mel Frequency Cepstrum coefficient (MFCC) and Linear Prediction Cepstrum coefficient (LPCC) are taken in use for feature extraction from the provided voice sample. This paper provides a comparative study of MFCC and LPCC based on the accuracy of results and their working methodology. The results are better if MFCC is used for feature extraction.
OPTIMIZING VOIP USING A CROSS LAYER CALL ADMISSION CONTROL SCHEMEIJCNCJournal
This document discusses optimizing VoIP quality over wireless networks using a cross-layer call admission control scheme. It proposes monitoring real-time control protocol reports and data rates at the MAC layer to determine when quality is degraded. When quality degrades due to issues like network congestion or variable transmission rates, the solution is to adapt the packet size or codec type. The proposed scheme is simulated using a wireless campus network model to improve performance.
LPC Models and Different Speech Enhancement Techniques- A Reviewijiert bestjournal
Author has already published one review paper on the quality enhancement of a speech signal by minimizing the noise. This is a second paper of same series. In last two decades the researchers have taken continuous efforts to reduce the noise signal from the speech signal. Th is paper comments on,various study carried out and analysis propos als of the researchers for en hancement of the quality of speech signal. Various models,coding,speech quality improvement methods,speaker dependent codebooks,autocorrelation subtraction,speech restoration,producing speech at low bit rates,compression and enhancement are the vari ous aspects of speech enhancement. We have presented the review of all above mentioned technologies in this paper and also willing to examine few of the techniques in order to analyze the factors affecting them in upcoming paper of the series.
An Empirical Evaluation of VoIP Playout Buffer Dimensioning in Skype, Google ...Academia Sinica
VoIP playout buffer dimensioning has long been a challeng- ing optimization problem, as the buffer size must maintain a balance between conversational interactivity and speech quality. The conversational quality may be affected by a number of factors, some of which may change over time. Although a great deal of research effort has been expended in trying to solve the problem, how the research results are applied in practice is unclear.
In this paper, we investigate the playout buffer dimension- ing algorithms applied in three popular VoIP applications, namely, Skype, Google Talk, and MSN Messenger. We conduct experiments to assess how the applications adjust their playout buffer sizes. Using an objective QoE (Quality of Experience) metric, we show that Google Talk and MSN Messenger do not adjust their respective buffer sizes appropriately, while Skype does not adjust its buffer at all. In other words, they could provide better QoE to users by improving their buffer dimensioning algorithms. Moreover, none of the applications adapts its buffer size to the network loss rate, which should also be considered to ensure optimal QoE provisioning.
The document discusses research on quantifying user satisfaction (QoE) in VoIP applications like Skype calls. It presents three key contributions:
1) Developing the first QoE measurement methodology based on analyzing large-scale Skype call data to correlate call duration with network quality factors like jitter and bit rate.
2) Proposing OneClick, a simple framework for crowdsourced QoE experiments based on user clicks to indicate dissatisfaction.
3) Introducing the first crowdsourcable QoE evaluation methodology to verify user judgments.
Highlights of ITU-T Study Group 12 meeting in Jan 2017 : Performance, Qualit...ITU
Highlihgts of the results and outcomes of ITU-T Study Group 12 meeting in January 2017 : Performance, Quality of Service and Quality of Experience
Author : Kwame BAAH-ACHEAMFUOR, Study Group 12 Chairman
An Evolutionary Approach to Speech Quality Estimationadil raja
This document presents research using genetic programming to develop non-intrusive models for estimating voice over IP (VoIP) quality. Researchers used a VoIP simulation environment to generate distorted speech files under different network conditions and trained genetic programs to map transport layer metrics like packet loss and delay to mean opinion scores. The best models achieved good accuracy compared to the intrusive PESQ standard with only 1-3 variables, making them suitable for real-time VoIP quality monitoring. Future work aims to include wideband codecs and develop a unified quality estimation model.
The document discusses methods for objective and subjective video quality assessment and speech enhancement. It covers four parts: (1) a classification and review of no-reference visual quality assessment methods, (2) no-reference and reduced-reference methods for video quality assessment including neural network and support vector machine approaches, (3) subjective methods for video quality assessment including studies on low resolution videos and crowdsourcing, and (4) speech enhancement techniques including spectral center-of-gravity based demodulation and convex optimization based demodulation. The document evaluates various computational models and machine learning techniques for video and speech quality assessment.
05 comparative study of voice print based acoustic features mfcc and lpccIJAEMSJORNAL
Voice is the best biometric feature for investigation and authentication. It has both biological and behavioural features. The acoustic features are related to the voice. The Speaker Recognition System is designed for the automatic authentication of speaker’s identity which is truly based on the human’s voice. Mel Frequency Cepstrum coefficient (MFCC) and Linear Prediction Cepstrum coefficient (LPCC) are taken in use for feature extraction from the provided voice sample. This paper provides a comparative study of MFCC and LPCC based on the accuracy of results and their working methodology. The results are better if MFCC is used for feature extraction.
OPTIMIZING VOIP USING A CROSS LAYER CALL ADMISSION CONTROL SCHEMEIJCNCJournal
This document discusses optimizing VoIP quality over wireless networks using a cross-layer call admission control scheme. It proposes monitoring real-time control protocol reports and data rates at the MAC layer to determine when quality is degraded. When quality degrades due to issues like network congestion or variable transmission rates, the solution is to adapt the packet size or codec type. The proposed scheme is simulated using a wireless campus network model to improve performance.
LPC Models and Different Speech Enhancement Techniques- A Reviewijiert bestjournal
Author has already published one review paper on the quality enhancement of a speech signal by minimizing the noise. This is a second paper of same series. In last two decades the researchers have taken continuous efforts to reduce the noise signal from the speech signal. Th is paper comments on,various study carried out and analysis propos als of the researchers for en hancement of the quality of speech signal. Various models,coding,speech quality improvement methods,speaker dependent codebooks,autocorrelation subtraction,speech restoration,producing speech at low bit rates,compression and enhancement are the vari ous aspects of speech enhancement. We have presented the review of all above mentioned technologies in this paper and also willing to examine few of the techniques in order to analyze the factors affecting them in upcoming paper of the series.
An Empirical Evaluation of VoIP Playout Buffer Dimensioning in Skype, Google ...Academia Sinica
VoIP playout buffer dimensioning has long been a challeng- ing optimization problem, as the buffer size must maintain a balance between conversational interactivity and speech quality. The conversational quality may be affected by a number of factors, some of which may change over time. Although a great deal of research effort has been expended in trying to solve the problem, how the research results are applied in practice is unclear.
In this paper, we investigate the playout buffer dimension- ing algorithms applied in three popular VoIP applications, namely, Skype, Google Talk, and MSN Messenger. We conduct experiments to assess how the applications adjust their playout buffer sizes. Using an objective QoE (Quality of Experience) metric, we show that Google Talk and MSN Messenger do not adjust their respective buffer sizes appropriately, while Skype does not adjust its buffer at all. In other words, they could provide better QoE to users by improving their buffer dimensioning algorithms. Moreover, none of the applications adapts its buffer size to the network loss rate, which should also be considered to ensure optimal QoE provisioning.
The document discusses research on quantifying user satisfaction (QoE) in VoIP applications like Skype calls. It presents three key contributions:
1) Developing the first QoE measurement methodology based on analyzing large-scale Skype call data to correlate call duration with network quality factors like jitter and bit rate.
2) Proposing OneClick, a simple framework for crowdsourced QoE experiments based on user clicks to indicate dissatisfaction.
3) Introducing the first crowdsourcable QoE evaluation methodology to verify user judgments.
Highlights of ITU-T Study Group 12 meeting in Jan 2017 : Performance, Qualit...ITU
Highlihgts of the results and outcomes of ITU-T Study Group 12 meeting in January 2017 : Performance, Quality of Service and Quality of Experience
Author : Kwame BAAH-ACHEAMFUOR, Study Group 12 Chairman
DEVELOPMENT OF SPEAKER VERIFICATION UNDER LIMITED DATA AND CONDITIONniranjan kumar
1) The document presents a literature review and proposal for a speaker verification system using voice passwords under guidance of Dr. G. Pradhan at NIT Patna.
2) The goals are to develop a text-independent speaker verification system that can perform well with short speech samples in limited data conditions by modeling speaker information and reducing phonetic variability.
3) Baseline experiments use MFCC features with GMM modeling, achieving better performance with more Gaussian mixtures; future work will explore other features and modeling techniques to improve robustness to mismatches.
Real-Time Non-Intrusive Speech Quality Estimation for VoIPadil raja
This document discusses research on developing a real-time, non-intrusive model for estimating speech quality over VoIP networks. The research aims to assess speech quality at mid-network points by evaluating both transport layer metrics like packet loss and jitter, as well as speech layer metrics using perceptual models. The current status describes capturing transport metrics from RTP/RTCP and implementing a perceptual model on an IXP2400 processor. Future work involves integrating the transport and speech models, testing under various network conditions, and evaluating the model for multiple calls and low bitrate codecs.
A Distortion-Resistant Routing Framework for Video Traffic in Wireless Multih...1crore projects
IEEE PROJECTS 2015
1 crore projects is a leading Guide for ieee Projects and real time projects Works Provider.
It has been provided Lot of Guidance for Thousands of Students & made them more beneficial in all Technology Training.
Dot Net
DOTNET Project Domain list 2015
1. IEEE based on datamining and knowledge engineering
2. IEEE based on mobile computing
3. IEEE based on networking
4. IEEE based on Image processing
5. IEEE based on Multimedia
6. IEEE based on Network security
7. IEEE based on parallel and distributed systems
Java Project Domain list 2015
1. IEEE based on datamining and knowledge engineering
2. IEEE based on mobile computing
3. IEEE based on networking
4. IEEE based on Image processing
5. IEEE based on Multimedia
6. IEEE based on Network security
7. IEEE based on parallel and distributed systems
ECE IEEE Projects 2015
1. Matlab project
2. Ns2 project
3. Embedded project
4. Robotics project
Eligibility
Final Year students of
1. BSc (C.S)
2. BCA/B.E(C.S)
3. B.Tech IT
4. BE (C.S)
5. MSc (C.S)
6. MSc (IT)
7. MCA
8. MS (IT)
9. ME(ALL)
10. BE(ECE)(EEE)(E&I)
TECHNOLOGY USED AND FOR TRAINING IN
1. DOT NET
2. C sharp
3. ASP
4. VB
5. SQL SERVER
6. JAVA
7. J2EE
8. STRINGS
9. ORACLE
10. VB dotNET
11. EMBEDDED
12. MAT LAB
13. LAB VIEW
14. Multi Sim
CONTACT US
1 CRORE PROJECTS
Door No: 214/215,2nd Floor,
No. 172, Raahat Plaza, (Shopping Mall) ,Arcot Road, Vadapalani, Chennai,
Tamin Nadu, INDIA - 600 026
Email id: 1croreprojects@gmail.com
website:1croreprojects.com
Phone : +91 97518 00789 / +91 72999 51536
Higher Order Low Pass FIR Filter Design using IPSO Algorithmijtsrd
This paper presents an optimal design of digital low pass finite impulse response FIR filter using Improved Particle Swarm Optimization IPSO . The design target of FIR filter is to approximate the ideal filters on the request of a given designing specifications. The traditional based optimization techniques are not efficient for digital filter design. The filter specification to be realized IPSO algorithm generates the best coefficients and try to meet the ideal frequency response. Improved Particle swarm optimization PSO proposes a new equation for the velocity vector and updating the particle vectors and hence the solution quality is improved. The IPSO technique enhances its search capability that leads to a higher probability of obtaining the optimal solution. In this paper for the given problem the realization of the FIR filter has been performed. The simulation results have been performed by using the improved particle swarm optimization IPSO method. M. Santhanaraj | Rishikesh. S. S | Subramanian. A. N | Vijai Sooriya. Su ""Higher Order Low Pass FIR Filter Design using IPSO Algorithm"" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-3 , April 2019, URL: https://www.ijtsrd.com/papers/ijtsrd22899.pdf
Paper URL: https://www.ijtsrd.com/engineering/electronics-and-communication-engineering/22899/higher-order-low-pass-fir-filter-design-using-ipso-algorithm/m-santhanaraj
The success of Skype has inspired a generation of peer-to-peer-based solutions for satisfactory real-time multimedia services over the Internet. However, fundamental questions, such as whether VoIP services like Skype are good enough in terms of user satisfaction, have not been formally addressed. One of the major challenges lies in the lack of an easily accessible and objective index to quantify the degree of user satisfaction.
In this work, we propose a model, geared to Skype, but generalizable to other VoIP services, to quantify VoIP user satisfaction based on a rigorous analysis of the call duration from actual Skype traces. The User Satisfaction Index (USI) derived from the model is unique in that 1) it is composed by objective source- and network-level metrics, such as the bit rate, bit rate jitter, and round-trip time, 2) unlike speech quality measures based on voice signals, such as the PESQ model standardized by ITU-T, the metrics are easily accessible and computable for real-time adaptation, and 3) the model development only requires network measurements, i.e., no user surveys or voice signals are necessary. Our model is validated by an independent set of metrics that quantifies the degree of user interaction from the actual traces.
This document outlines a proposed study on developing a voice password-based speaker verification system. It will explore methods for modeling speakers with limited data, such as artificially generating multiple utterances from short speech segments. The study aims to reduce phonetic variability between training and test data. It will also examine score normalization techniques, comparing cohort-centric normalization typically used to a proposed speaker-centric approach. The goal is to build a text-independent voice password system that can reliably verify identities from short speech samples, improving security and enabling remote access applications.
IRJET- Survey on Efficient Signal Processing Techniques for Speech EnhancementIRJET Journal
This document provides a survey of various speech enhancement techniques. It discusses five papers that propose different speech enhancement algorithms: 1) Discrete Tchebichef Transform and Discrete Krawtchouk Transform for removing noise using minimum mean square error. 2) Empirical mode decomposition and adaptive centre weighted average filtering that is effective for removing noise components. 3) Adaptive Wiener filtering that adapts the filter transfer function based on speech signal statistics. 4) Compressive sensing based speech enhancement that handles non-sparse noise. 5) Wavelet packet transform and non-negative matrix factorization to emphasize the speech components in each sub-band. The document also discusses speech enhancement using deep neural networks, empirical mode decomposition with Hurst exponent
This document describes research on improving the naturalness and intelligibility of alaryngeal speech using voice conversion and synthetic fundamental frequency (F0). The researchers used conditional generative adversarial networks (cGANs) to perform voice conversion from alaryngeal speech to clearer laryngeal speech. They also developed methods to predict voicing, degree of voicing, and speech spectra from alaryngeal speech features and synthesize pitch accent curves. Subjective evaluations found that the combined use of voice conversion and F0 synthesis increased the naturalness of alaryngeal speech compared to the original. The goal of the research was to automatically improve the quality of speech for individuals who have undergone total laryngectomy.
VIDEO QUALITY ASSESSMENT USING LAPLACIAN MODELING OF MOTION VECTOR DISTRIBUTI...sipij
Video/Image quality assessment (VQA/IQA) is fundamental in various fields of video/image processing.
VQA reflects the quality of a video as most people commonly perceive. This paper proposes a reducedreference
mobile VQA, in which one-dimensional (1-D) motion vector (MV) distributions are used as
features of videos. This paper focuses on reduction of data size using Laplacian modeling of MV
distributions because network resource is restricted in the case of mobile video. The proposed method is
more efficient than the conventional methods in view of the computation time, because the proposed quality
metric decodes MVs directly from video stream in the parsing process rather than reconstructing the
distorted video at a receiver. Moreover, in view of data size, the proposed method is efficient because a
sender transmits only 28 parameters. We adopt the Laplacian distribution for modeling 1-D MV
histograms. 1-D MV histograms accumulated over the whole video sequences are used, which is different
from the conventional methods that assess each image frame independently. For testing the similarity
between MV histogram of reference and distorted videos and for minimizing the fitting error in Laplacian
modeling process, we use the chi-square method. To show the effectiveness of our proposed method, we
compare the proposed method with the conventional methods with coded video clips, which are coded
under varying bit rate, image size, and frame rate by H.263 and H.264/AVC. Experimental results show
that the proposed method gives the performance comparable with the conventional methods, especially, the
proposed method requires much lower transmission data.
Real-time Video Quality Assessment for Analog Television Based on Adaptive Fu...TELKOMNIKA JOURNAL
Real-time VQA (Video Quality Assessment) is an important part in the effort to build tracking
antenna system especially for analog TV. In this case, VQA must work in real-time to assess the video
clarity level. VQA assessment results are valuable information for the decision-making process. Thus, the
antenna can rotate automatically looking for the ideal direction without user’s control. In addition, the video
clarity level on the TV screen can reach optimum according to the user's wishes. The biggest challenge to
VQA is, VQA must be able to assess the video clarity level according to the user’s visual perception.
Therefore, in this study, the MOS-VQS (Mean Opinion Score-Video Quality Subjective) was used as a
visual perception approach. In addition, Adaptive FIS (Fuzzy Inference System) with membership function
tuning was implemented for decision making. This was conducted as an effort to build a reliable real-time
VQA. The test results show that real-time VQA that has been built has a good performance. This is shown
from the average accuracy percentage of the lowest assessment reached 77.2% and the highest reached
88.2%.
This document provides an overview of the ACM workshop on advanced video streaming techniques for peer-to-peer networks and social networking. It summarizes the topics of interest for the workshop, which included innovative P2P video streaming solutions, social media content distribution, and advanced video coding techniques for real-time applications. It also summarizes the 15 papers accepted to the workshop, which covered topics like multi-source video distribution, modeling end-to-end delay, and improvements to quality of experience for multiple description video transmission. An invited talk was also given by a representative from the BBC on audio/visual content delivery over P2P networks.
Voice over IP (VoIP) Speech Quality Measurement with Open-Source Software Com...Sebastian Schumann
This paper proposes an alternative to expensive means for VoIP speech quality measurement. While current applications and measurement devices on the market are very expensive, the authors propose a solution based on open-source components that allows the determination of the Mean Opinion Score (MOS) value according the Perceptual Evaluation of Speech Quality (PESQ) test methodology. Presented at Elmar 2010 in Zadar, Croatia.
Interactive voice conversion for augmented speech productionNU_I_TODALAB
This document discusses recent progress in interactive voice conversion techniques for augmenting speech production. It begins by explaining the physical limitations of normal speech production and how voice conversion can augment speech by controlling more information. It then discusses how interactive voice conversion allows for quick response times, better controllability through real-time feedback, and understanding user intent from multimodal behavior signals. Recent advances discussed include low-latency voice conversion networks, controllable waveform generation respecting the source-filter model of speech, and expression control using signals like arm movements. The goal is to develop cooperatively augmented speech that can help users with lost speech abilities.
Robust audio watermarking based on transform domain and SVD with compressive ...TELKOMNIKA JOURNAL
The growth of the internet and digital data has resulted forgery, modification and sharing of digital data without property rights. Audio watermarking is one of a solution to protect the copyright of an audio from copyright infringement. This paper proposes an audio watermarking method which is robust against attacks and high capacity. First, a synchronization bit is added to the audio host. After the audio host is decomposed by Lifting Wavelet Transform (LWT), then choose a subband from the output of LWT to be transformed by discrete cosine transform (DCT). Next, the matrix of the signal from DCT is selected for the singular value decomposition (SVD) process, so that is obtained U, S and V matrix. S matrix is embedded with the watermark. Before the embedding process, the watermark image is compressed by Compressive Sampling. The results show that the proposed watermarking system is highly robust against a kind attack of LPF, resampling, and linear speed change which is proven by its BER is zero.
This document discusses feature extraction techniques for isolated word speech recognition. It begins with an introduction to digital speech processing and speech recognition models. The main part of the document compares two common feature extraction techniques: Mel Frequency Cepstral Coefficients (MFCC) and Relative Spectral (RASTA) filtering. MFCC allows signals to extract feature vectors and provides high performance but lacks robustness. RASTA filtering reduces the impact of noise in signals and provides high robustness by band-passing feature coefficients in both log spectral and spectral domains. The document provides details on the process of MFCC feature extraction, which involves steps like framing, windowing, fast Fourier transform, mel filtering, discrete cosine transform, and calculating
This document analyzes the performance of a four user optical code division multiple access (OCDMA) communication system under the effect of jitter. Simulations were conducted using Rsoft Optsim to evaluate the bit error rate (BER) and Q-factor of the system with jitter varying from 0 to 3 picoseconds. The results show that as jitter increases, BER increases and Q-factor decreases for different fiber lengths of 10km, 30km, 50km and 70km. Specifically, BER varies from 2.11E-02 to 1.71E-02 as jitter increases for a 10km fiber length. For a 30km length, BER increases from 2.11E-02 to 1.71
Text Prompted Remote Speaker Authentication : Joint Speech and Speaker Recogn...gt_ebuddy
Joint Speech and Speaker Recognition using Hidden Markov Model/Vector Quantization for speaker independent Speech Recognition and Gaussian Mixture Model for speech independent speaker recognition- used MFCC (Mel-Frequency Cepstral Coefficient) for Feature Extraction (delta,delta delta and energy - 39 coefficients).
Developed in JAVA with client/server Architecture, web interface developed in Adobe Flex.
This project was done at TU, IOE - Pulchowk Campus, Nepal.
For more details visit http://ganeshtiwaridotcomdotnp.blogspot.com
ABSTRACT OF PROJECT>>>
Biometric is physical characteristic unique to each individual. It has a very useful application in authentication and access control.
The designed system is a text-prompted version of voice biometric which incorporates text-independent speaker verification and speaker-independent speech verification system implemented independently. The foundation for this joint system is that the speech signal conveys both the speech content and speaker identity. Such systems are more-secure from playback attack, since the word to speak during authentication is not previously set.
During the course of the project various digital signal processing and pattern classification algorithms were studied. Short time spectral analysis was performed to obtain MFCC, energy and their deltas as feature. Feature extraction module is same for both systems. Speaker modeling was done by GMM and Left to Right Discrete HMM with VQ was used for isolated word modeling. And results of both systems were combined to authenticate the user.
The speech model for each word was pre-trained by using utterance of 45 English words. The speaker model was trained by utterance of about 2 minutes each by 15 speakers. While uttering the individual words, the recognition rate of the speech recognition system is 92 % and speaker recognition system is 66%. For longer duration of utterance (>5sec) the recognition rate of speaker recognition system improves to 78%.
Audio Steganography Coding Using the Discreet Wavelet TransformsCSCJournals
The performance of audio steganography compression system using discreet wavelet transform (DWT) is investigated. Audio steganography coding is the technology of transforming stego-speech into efficiently encoded version that can be decoded in the receiver side to produce a close representation of the initial signal (non compressed). Experimental results prove the efficiency of the used compression technique since the compressed stego-speech are perceptually intelligible and indistinguishable from the equivalent initial signal, while being able to recover the initial stego-speech with slight degradation in the quality .
There is a massive growth in mobile video consumption which outpaces the capacity improvements in next generation mobile networks. Specifically, mobile network operators face the challenge of allocating the scarce wireless resources while maximizing the user quality of experience (QoE). The first part of this talk addresses the main challenges in uplink distribution of user-generated video content over fourth generation mobile networks. The second part explores the benefit of QoE-based traffic and resource management in the mobile network in the context of adaptive HTTP downlink video delivery.
This document provides an overview of design patterns, which are proven solutions to common problems in software design. It discusses different types of patterns like creational, structural, and behavioral patterns. It then gives examples of some common patterns like Iterator, Strategy, and Factory Method. The Factory Method pattern allows defining an interface for creating objects but letting subclasses decide which objects to instantiate. The Strategy pattern defines a family of algorithms, puts each of them in a separate class, and makes their objects interchangeable.
The document discusses different types of polymorphic variables in object-oriented programming. It describes the receiver variable, which holds the object instance being operated on during method calls. Reverse polymorphism is the process of downcasting a polymorphic variable to access methods of a more specific type. Pure polymorphism occurs when a polymorphic variable is passed as a method argument, allowing different behavior based on the variable's underlying type.
The document describes file transfer between a server and client using sockets in Java. The server code opens a server socket to listen for connections, reads files from the client, and sends the file content back to the client. The client code connects to the server, sends a file name, receives the file content from the server and writes it to a local file. The document includes code snippets for both the server and client implementations.
DEVELOPMENT OF SPEAKER VERIFICATION UNDER LIMITED DATA AND CONDITIONniranjan kumar
1) The document presents a literature review and proposal for a speaker verification system using voice passwords under guidance of Dr. G. Pradhan at NIT Patna.
2) The goals are to develop a text-independent speaker verification system that can perform well with short speech samples in limited data conditions by modeling speaker information and reducing phonetic variability.
3) Baseline experiments use MFCC features with GMM modeling, achieving better performance with more Gaussian mixtures; future work will explore other features and modeling techniques to improve robustness to mismatches.
Real-Time Non-Intrusive Speech Quality Estimation for VoIPadil raja
This document discusses research on developing a real-time, non-intrusive model for estimating speech quality over VoIP networks. The research aims to assess speech quality at mid-network points by evaluating both transport layer metrics like packet loss and jitter, as well as speech layer metrics using perceptual models. The current status describes capturing transport metrics from RTP/RTCP and implementing a perceptual model on an IXP2400 processor. Future work involves integrating the transport and speech models, testing under various network conditions, and evaluating the model for multiple calls and low bitrate codecs.
A Distortion-Resistant Routing Framework for Video Traffic in Wireless Multih...1crore projects
IEEE PROJECTS 2015
1 crore projects is a leading Guide for ieee Projects and real time projects Works Provider.
It has been provided Lot of Guidance for Thousands of Students & made them more beneficial in all Technology Training.
Dot Net
DOTNET Project Domain list 2015
1. IEEE based on datamining and knowledge engineering
2. IEEE based on mobile computing
3. IEEE based on networking
4. IEEE based on Image processing
5. IEEE based on Multimedia
6. IEEE based on Network security
7. IEEE based on parallel and distributed systems
Java Project Domain list 2015
1. IEEE based on datamining and knowledge engineering
2. IEEE based on mobile computing
3. IEEE based on networking
4. IEEE based on Image processing
5. IEEE based on Multimedia
6. IEEE based on Network security
7. IEEE based on parallel and distributed systems
ECE IEEE Projects 2015
1. Matlab project
2. Ns2 project
3. Embedded project
4. Robotics project
Eligibility
Final Year students of
1. BSc (C.S)
2. BCA/B.E(C.S)
3. B.Tech IT
4. BE (C.S)
5. MSc (C.S)
6. MSc (IT)
7. MCA
8. MS (IT)
9. ME(ALL)
10. BE(ECE)(EEE)(E&I)
TECHNOLOGY USED AND FOR TRAINING IN
1. DOT NET
2. C sharp
3. ASP
4. VB
5. SQL SERVER
6. JAVA
7. J2EE
8. STRINGS
9. ORACLE
10. VB dotNET
11. EMBEDDED
12. MAT LAB
13. LAB VIEW
14. Multi Sim
CONTACT US
1 CRORE PROJECTS
Door No: 214/215,2nd Floor,
No. 172, Raahat Plaza, (Shopping Mall) ,Arcot Road, Vadapalani, Chennai,
Tamin Nadu, INDIA - 600 026
Email id: 1croreprojects@gmail.com
website:1croreprojects.com
Phone : +91 97518 00789 / +91 72999 51536
Higher Order Low Pass FIR Filter Design using IPSO Algorithmijtsrd
This paper presents an optimal design of digital low pass finite impulse response FIR filter using Improved Particle Swarm Optimization IPSO . The design target of FIR filter is to approximate the ideal filters on the request of a given designing specifications. The traditional based optimization techniques are not efficient for digital filter design. The filter specification to be realized IPSO algorithm generates the best coefficients and try to meet the ideal frequency response. Improved Particle swarm optimization PSO proposes a new equation for the velocity vector and updating the particle vectors and hence the solution quality is improved. The IPSO technique enhances its search capability that leads to a higher probability of obtaining the optimal solution. In this paper for the given problem the realization of the FIR filter has been performed. The simulation results have been performed by using the improved particle swarm optimization IPSO method. M. Santhanaraj | Rishikesh. S. S | Subramanian. A. N | Vijai Sooriya. Su ""Higher Order Low Pass FIR Filter Design using IPSO Algorithm"" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-3 , April 2019, URL: https://www.ijtsrd.com/papers/ijtsrd22899.pdf
Paper URL: https://www.ijtsrd.com/engineering/electronics-and-communication-engineering/22899/higher-order-low-pass-fir-filter-design-using-ipso-algorithm/m-santhanaraj
The success of Skype has inspired a generation of peer-to-peer-based solutions for satisfactory real-time multimedia services over the Internet. However, fundamental questions, such as whether VoIP services like Skype are good enough in terms of user satisfaction, have not been formally addressed. One of the major challenges lies in the lack of an easily accessible and objective index to quantify the degree of user satisfaction.
In this work, we propose a model, geared to Skype, but generalizable to other VoIP services, to quantify VoIP user satisfaction based on a rigorous analysis of the call duration from actual Skype traces. The User Satisfaction Index (USI) derived from the model is unique in that 1) it is composed by objective source- and network-level metrics, such as the bit rate, bit rate jitter, and round-trip time, 2) unlike speech quality measures based on voice signals, such as the PESQ model standardized by ITU-T, the metrics are easily accessible and computable for real-time adaptation, and 3) the model development only requires network measurements, i.e., no user surveys or voice signals are necessary. Our model is validated by an independent set of metrics that quantifies the degree of user interaction from the actual traces.
This document outlines a proposed study on developing a voice password-based speaker verification system. It will explore methods for modeling speakers with limited data, such as artificially generating multiple utterances from short speech segments. The study aims to reduce phonetic variability between training and test data. It will also examine score normalization techniques, comparing cohort-centric normalization typically used to a proposed speaker-centric approach. The goal is to build a text-independent voice password system that can reliably verify identities from short speech samples, improving security and enabling remote access applications.
IRJET- Survey on Efficient Signal Processing Techniques for Speech EnhancementIRJET Journal
This document provides a survey of various speech enhancement techniques. It discusses five papers that propose different speech enhancement algorithms: 1) Discrete Tchebichef Transform and Discrete Krawtchouk Transform for removing noise using minimum mean square error. 2) Empirical mode decomposition and adaptive centre weighted average filtering that is effective for removing noise components. 3) Adaptive Wiener filtering that adapts the filter transfer function based on speech signal statistics. 4) Compressive sensing based speech enhancement that handles non-sparse noise. 5) Wavelet packet transform and non-negative matrix factorization to emphasize the speech components in each sub-band. The document also discusses speech enhancement using deep neural networks, empirical mode decomposition with Hurst exponent
This document describes research on improving the naturalness and intelligibility of alaryngeal speech using voice conversion and synthetic fundamental frequency (F0). The researchers used conditional generative adversarial networks (cGANs) to perform voice conversion from alaryngeal speech to clearer laryngeal speech. They also developed methods to predict voicing, degree of voicing, and speech spectra from alaryngeal speech features and synthesize pitch accent curves. Subjective evaluations found that the combined use of voice conversion and F0 synthesis increased the naturalness of alaryngeal speech compared to the original. The goal of the research was to automatically improve the quality of speech for individuals who have undergone total laryngectomy.
VIDEO QUALITY ASSESSMENT USING LAPLACIAN MODELING OF MOTION VECTOR DISTRIBUTI...sipij
Video/Image quality assessment (VQA/IQA) is fundamental in various fields of video/image processing.
VQA reflects the quality of a video as most people commonly perceive. This paper proposes a reducedreference
mobile VQA, in which one-dimensional (1-D) motion vector (MV) distributions are used as
features of videos. This paper focuses on reduction of data size using Laplacian modeling of MV
distributions because network resource is restricted in the case of mobile video. The proposed method is
more efficient than the conventional methods in view of the computation time, because the proposed quality
metric decodes MVs directly from video stream in the parsing process rather than reconstructing the
distorted video at a receiver. Moreover, in view of data size, the proposed method is efficient because a
sender transmits only 28 parameters. We adopt the Laplacian distribution for modeling 1-D MV
histograms. 1-D MV histograms accumulated over the whole video sequences are used, which is different
from the conventional methods that assess each image frame independently. For testing the similarity
between MV histogram of reference and distorted videos and for minimizing the fitting error in Laplacian
modeling process, we use the chi-square method. To show the effectiveness of our proposed method, we
compare the proposed method with the conventional methods with coded video clips, which are coded
under varying bit rate, image size, and frame rate by H.263 and H.264/AVC. Experimental results show
that the proposed method gives the performance comparable with the conventional methods, especially, the
proposed method requires much lower transmission data.
Real-time Video Quality Assessment for Analog Television Based on Adaptive Fu...TELKOMNIKA JOURNAL
Real-time VQA (Video Quality Assessment) is an important part in the effort to build tracking
antenna system especially for analog TV. In this case, VQA must work in real-time to assess the video
clarity level. VQA assessment results are valuable information for the decision-making process. Thus, the
antenna can rotate automatically looking for the ideal direction without user’s control. In addition, the video
clarity level on the TV screen can reach optimum according to the user's wishes. The biggest challenge to
VQA is, VQA must be able to assess the video clarity level according to the user’s visual perception.
Therefore, in this study, the MOS-VQS (Mean Opinion Score-Video Quality Subjective) was used as a
visual perception approach. In addition, Adaptive FIS (Fuzzy Inference System) with membership function
tuning was implemented for decision making. This was conducted as an effort to build a reliable real-time
VQA. The test results show that real-time VQA that has been built has a good performance. This is shown
from the average accuracy percentage of the lowest assessment reached 77.2% and the highest reached
88.2%.
This document provides an overview of the ACM workshop on advanced video streaming techniques for peer-to-peer networks and social networking. It summarizes the topics of interest for the workshop, which included innovative P2P video streaming solutions, social media content distribution, and advanced video coding techniques for real-time applications. It also summarizes the 15 papers accepted to the workshop, which covered topics like multi-source video distribution, modeling end-to-end delay, and improvements to quality of experience for multiple description video transmission. An invited talk was also given by a representative from the BBC on audio/visual content delivery over P2P networks.
Voice over IP (VoIP) Speech Quality Measurement with Open-Source Software Com...Sebastian Schumann
This paper proposes an alternative to expensive means for VoIP speech quality measurement. While current applications and measurement devices on the market are very expensive, the authors propose a solution based on open-source components that allows the determination of the Mean Opinion Score (MOS) value according the Perceptual Evaluation of Speech Quality (PESQ) test methodology. Presented at Elmar 2010 in Zadar, Croatia.
Interactive voice conversion for augmented speech productionNU_I_TODALAB
This document discusses recent progress in interactive voice conversion techniques for augmenting speech production. It begins by explaining the physical limitations of normal speech production and how voice conversion can augment speech by controlling more information. It then discusses how interactive voice conversion allows for quick response times, better controllability through real-time feedback, and understanding user intent from multimodal behavior signals. Recent advances discussed include low-latency voice conversion networks, controllable waveform generation respecting the source-filter model of speech, and expression control using signals like arm movements. The goal is to develop cooperatively augmented speech that can help users with lost speech abilities.
Robust audio watermarking based on transform domain and SVD with compressive ...TELKOMNIKA JOURNAL
The growth of the internet and digital data has resulted forgery, modification and sharing of digital data without property rights. Audio watermarking is one of a solution to protect the copyright of an audio from copyright infringement. This paper proposes an audio watermarking method which is robust against attacks and high capacity. First, a synchronization bit is added to the audio host. After the audio host is decomposed by Lifting Wavelet Transform (LWT), then choose a subband from the output of LWT to be transformed by discrete cosine transform (DCT). Next, the matrix of the signal from DCT is selected for the singular value decomposition (SVD) process, so that is obtained U, S and V matrix. S matrix is embedded with the watermark. Before the embedding process, the watermark image is compressed by Compressive Sampling. The results show that the proposed watermarking system is highly robust against a kind attack of LPF, resampling, and linear speed change which is proven by its BER is zero.
This document discusses feature extraction techniques for isolated word speech recognition. It begins with an introduction to digital speech processing and speech recognition models. The main part of the document compares two common feature extraction techniques: Mel Frequency Cepstral Coefficients (MFCC) and Relative Spectral (RASTA) filtering. MFCC allows signals to extract feature vectors and provides high performance but lacks robustness. RASTA filtering reduces the impact of noise in signals and provides high robustness by band-passing feature coefficients in both log spectral and spectral domains. The document provides details on the process of MFCC feature extraction, which involves steps like framing, windowing, fast Fourier transform, mel filtering, discrete cosine transform, and calculating
This document analyzes the performance of a four user optical code division multiple access (OCDMA) communication system under the effect of jitter. Simulations were conducted using Rsoft Optsim to evaluate the bit error rate (BER) and Q-factor of the system with jitter varying from 0 to 3 picoseconds. The results show that as jitter increases, BER increases and Q-factor decreases for different fiber lengths of 10km, 30km, 50km and 70km. Specifically, BER varies from 2.11E-02 to 1.71E-02 as jitter increases for a 10km fiber length. For a 30km length, BER increases from 2.11E-02 to 1.71
Text Prompted Remote Speaker Authentication : Joint Speech and Speaker Recogn...gt_ebuddy
Joint Speech and Speaker Recognition using Hidden Markov Model/Vector Quantization for speaker independent Speech Recognition and Gaussian Mixture Model for speech independent speaker recognition- used MFCC (Mel-Frequency Cepstral Coefficient) for Feature Extraction (delta,delta delta and energy - 39 coefficients).
Developed in JAVA with client/server Architecture, web interface developed in Adobe Flex.
This project was done at TU, IOE - Pulchowk Campus, Nepal.
For more details visit http://ganeshtiwaridotcomdotnp.blogspot.com
ABSTRACT OF PROJECT>>>
Biometric is physical characteristic unique to each individual. It has a very useful application in authentication and access control.
The designed system is a text-prompted version of voice biometric which incorporates text-independent speaker verification and speaker-independent speech verification system implemented independently. The foundation for this joint system is that the speech signal conveys both the speech content and speaker identity. Such systems are more-secure from playback attack, since the word to speak during authentication is not previously set.
During the course of the project various digital signal processing and pattern classification algorithms were studied. Short time spectral analysis was performed to obtain MFCC, energy and their deltas as feature. Feature extraction module is same for both systems. Speaker modeling was done by GMM and Left to Right Discrete HMM with VQ was used for isolated word modeling. And results of both systems were combined to authenticate the user.
The speech model for each word was pre-trained by using utterance of 45 English words. The speaker model was trained by utterance of about 2 minutes each by 15 speakers. While uttering the individual words, the recognition rate of the speech recognition system is 92 % and speaker recognition system is 66%. For longer duration of utterance (>5sec) the recognition rate of speaker recognition system improves to 78%.
Audio Steganography Coding Using the Discreet Wavelet TransformsCSCJournals
The performance of audio steganography compression system using discreet wavelet transform (DWT) is investigated. Audio steganography coding is the technology of transforming stego-speech into efficiently encoded version that can be decoded in the receiver side to produce a close representation of the initial signal (non compressed). Experimental results prove the efficiency of the used compression technique since the compressed stego-speech are perceptually intelligible and indistinguishable from the equivalent initial signal, while being able to recover the initial stego-speech with slight degradation in the quality .
There is a massive growth in mobile video consumption which outpaces the capacity improvements in next generation mobile networks. Specifically, mobile network operators face the challenge of allocating the scarce wireless resources while maximizing the user quality of experience (QoE). The first part of this talk addresses the main challenges in uplink distribution of user-generated video content over fourth generation mobile networks. The second part explores the benefit of QoE-based traffic and resource management in the mobile network in the context of adaptive HTTP downlink video delivery.
This document provides an overview of design patterns, which are proven solutions to common problems in software design. It discusses different types of patterns like creational, structural, and behavioral patterns. It then gives examples of some common patterns like Iterator, Strategy, and Factory Method. The Factory Method pattern allows defining an interface for creating objects but letting subclasses decide which objects to instantiate. The Strategy pattern defines a family of algorithms, puts each of them in a separate class, and makes their objects interchangeable.
The document discusses different types of polymorphic variables in object-oriented programming. It describes the receiver variable, which holds the object instance being operated on during method calls. Reverse polymorphism is the process of downcasting a polymorphic variable to access methods of a more specific type. Pure polymorphism occurs when a polymorphic variable is passed as a method argument, allowing different behavior based on the variable's underlying type.
The document describes file transfer between a server and client using sockets in Java. The server code opens a server socket to listen for connections, reads files from the client, and sends the file content back to the client. The client code connects to the server, sends a file name, receives the file content from the server and writes it to a local file. The document includes code snippets for both the server and client implementations.
The document discusses a wireless access research project involving automated file transfer between nodes. Shell scripts using SSH and SCP commands transfer files and execute commands securely between nodes. A GUI allows limited interactivity by invoking scripts to fetch results from remote nodes. The software represents physical nodes and their attributes as Node objects within an array. It also graphically displays the network topology based on routing tables, updating periodically.
This document provides formatting tips for writing assignments. It discusses using paragraphs, sentences, headings, and lists. For paragraphs, it recommends using proper spacing and writing smaller paragraphs. For sentences, it suggests writing shorter sentences. For headings, it advises using headings when important and keeping them small and capitalized. It also notes the two types of lists - bulleted and numbered - and recommends using numbered lists for a fixed number of points. The document cautions against using too much formatting like bold, italic, uppercase text.
A simple client-server application in java in which a client sends a message to a server and the server tries to be funny by sending back a funny response.
This document discusses the implications of substitution in object-oriented programming. It explores issues like memory allocation, the meaning of assignment, and differences between equality and identity testing. Key challenges include not knowing object sizes until runtime, which leads to complex semantics or dynamic objects and garbage collection. Dynamic semantics also tend toward pointer semantics for assignment and non-guarantees for equality. The programmer must be able to redefine equality as needed but this can introduce paradoxes.
Research Methods: Objectives and Contentsadil raja
This document outlines the objectives, contents, and conclusions of a research methods course. The objectives are to teach students basic writing, research, and presentation skills. The contents will start with simple writing exercises and progressively cover more advanced topics like reading and summarizing research papers from various disciplines. Examples of topics that may be covered include political science, marketing, and various optimization techniques. The document concludes by stating the goals are for students to learn in a relaxed way and efficiently apply their learning to get hired.
The document discusses data storage technologies used in computers. It covers RAM and ROM for temporary data storage, and hard disk drives for permanent mass storage of data. Hard disks use circular platters coated with magnetic material to store data in sectors and tracks accessed by read/write heads. Cache memory provides temporary high-speed storage. Buses like PCI are used to connect devices like video cards to the motherboard. The boot process loads the BIOS and boot loader from the hard disk to start the operating system.
An Evolutionary Approach to Speech Quality Estimation Using Genetic Programmingadil raja
This document presents research using genetic programming to develop non-intrusive models for estimating voice over IP (VoIP) call quality. The researchers used a VoIP simulation environment to generate distorted speech files under different network conditions. Genetic programming experiments were performed to evolve models relating mean opinion score (MOS) estimates to transport layer metrics like packet loss rate and jitter. The best models achieved mean squared errors within 0.04 of the intrusive PESQ algorithm estimates, performing significantly better than the ITU standard P.563 non-intrusive model. The models provide a simple, real-time alternative to PESQ for VoIP quality monitoring.
Inheritance allows a child class to inherit attributes and behaviors from a parent class. There are different forms of inheritance like specialization, where the child class is a specialized subtype of the parent, and specification inheritance where the parent class specifies behaviors implemented in the child class. Inheritance provides benefits like code reuse and reliability but also costs like increased complexity and slower execution speed.
This document outlines an object-oriented design approach called responsibility-driven design. It discusses key object-oriented design concepts like components, classes, instances, behavior and state. It also uses a sample interactive kitchen helper application to demonstrate design techniques like identifying components, creating CRC cards to define responsibilities and collaborators, and designing for change. The document is intended to guide readers through an object-oriented design process.
A Methodology for Deriving VoIP Equipment Impairment Factors for a Mixed NB/W...adil raja
This document discusses extending the E-model to account for mixed narrowband and wideband contexts in VoIP. It evaluates several methods for deriving impairment factors (Ie,eff) for wideband equipment based on PESQ scores. The best-fitting method maps PESQ scores from narrowband and wideband tests to derive a function relating Ie,wb,eff to factors like codec, packet loss, and delay. Scatter plots and surface plots show the derived relationships accurately estimate quality impairments in mixed narrowband and wideband networks.
This document discusses generics in C++. It introduces template functions and template classes as ways to reuse code by leaving key types unspecified. Template functions allow functions to operate on different data types, while template classes allow classes to work with different types. The document also discusses bounded genericity, which allows placing restrictions on template arguments, and issues with inheritance and generics.
This document contains source code for various components of a Java program that implements grammatical optimization. It includes the main class with the main method that initializes variables and objects. It also includes classes for the genotype, grammar, initializer, and mapper components. The classes define methods for initializing objects, mapping between genotypes and phenotypes, checking validity, and accessing attributes.
This document discusses different types of overloading in programming languages. It defines overloading as a term having multiple meanings resolved by context. There are two main contexts - scopes and type signatures. Overloading based on scopes allows the same name to be used in different classes or modules without ambiguity. Overloading based on type signatures resolves functions based on argument and return types. The document also discusses redefinitions in subclasses, optional parameters, and functions that take a variable number of arguments.
Xavor Pakistan was assessed for its CMM Level 3 capabilities. The assessment found that Xavor demonstrated Level 2 capabilities in requirements management, software project planning, software project tracking and oversight, software quality assurance, and software configuration management. Xavor demonstrated Level 3 capabilities in organization process focus, organization process definition, integrated software management, inter-group coordination, and peer reviews. The assessment provided recommendations for Xavor to improve its software configuration management processes, implement quantitative measurements, and conduct more rigorous statistical analysis and pilot projects to achieve Level 4 capabilities.
This document outlines a course about computer networks. It covers 9 major topics: introduction, the physical layer, data link layer, medium access layer, network layer, transport layer, application layer, network security, and conclusions. Each topic is further broken down into 3 or more subsections that will be taught as part of the course. The course aims to provide a comprehensive overview of computer network concepts and protocols.
Realtime, Non-Intrusive Evaluation of VoIP Using Genetic Programmingadil raja
The document discusses using genetic programming to develop a non-intrusive model for evaluating voice over IP (VoIP) quality based on transport layer network metrics. It first provides background on VoIP and challenges with speech quality assessment. It then describes a VoIP simulation environment and genetic programming approach. The goal is to evolve an estimation model for VoIP listening quality as a function of packet loss rate, jitter, bitrate and other transport metrics using genetic programming, and validate it against PESQ (Perceptual Evaluation of Speech Quality) scores.
Real-Time Non-Intrusive Speech Quality Estimation: A Signal-Based Modeladil raja
The document presents a non-intrusive speech quality estimation model developed using genetic programming. It discusses existing subjective and objective speech quality assessment methods. The proposed model uses genetic programming to derive a mapping from features extracted from speech signals using ITU-T P.563 to estimated mean opinion scores. The model outperforms the reference P.563 method with a 9.89% reduction in training error and 16.41% reduction in testing error. Key features selected by the model relate to vocal tract characteristics, distortion levels, and spectral properties.
Adaptive Optimization Schemes for Mobile VoIP Applications - Battery Life and...tumep
This document provides an abstract for a master's thesis that examines programmatic and application-layer methods for improving energy efficiency in mobile VoIP applications. The work focuses on optimizations suitable for VoIP implementations using SIP and IEEE 802.11 technologies. Energy-saving optimizations can impact perceived call quality, so energy-saving means are studied together with factors affecting call quality. Based on theory, adaptive optimization schemes for dynamically controlling an application's operation are proposed. A runtime quality model is developed for VoIP call quality estimation that can be integrated into optimization schemes. Power consumption measurements using the proposed optimization schemes show reductions in power usage can be achieved.
Syed Mohsin Ali is seeking a position as a 2G, 3G, and 4G Junior Optimizer where he can develop his technical and managerial skills. He has 3.5 years of experience in RF optimization working on projects for NSN, Huawei, and Etisalat UAE. His experience includes analyzing drive and walk tests to optimize LTE, 3G, and 2G networks to improve quality. He is proficient in using software like Nemo, Genex, and TEMS for tasks like troubleshooting coverage issues, monitoring KPIs, and generating optimization reports. He holds a B.E in Telecommunication Engineering from National University of Modern Languages.
Internet2: VoIP Phone Codec Testing White PaperJoshua Reola
This document summarizes a study that tested the effects of packet loss on voice quality for various Voice over IP (VoIP) codecs. Key findings include:
- The G.711 codec maintained acceptable audio quality (MOS >3.0) up to 5% packet loss, more than typically claimed by vendors.
- The G.729 codec degraded more quickly above 5% packet loss.
- The GIPS codec significantly outperformed standard codecs, maintaining quality up to 15% packet loss.
- There were no significant differences between vendors in codec performance. The study recommends the G.711 codec for networks with ample bandwidth and considering the GIPS codec. Further research on additional codecs and factors like packet jitter was advised
FutureComm 2010: Video Quality Analysis and MeasurementRADVISION Ltd.
This document summarizes RADVISION's video quality solutions. It discusses (1) factors that affect video quality like network conditions and content complexity, (2) RADVISION's video quality SDK and standalone testing suite that allow measuring and analyzing video quality in real-time, and (3) RADVISION's innovative research in areas like human vision databases and synchronization algorithms. The document emphasizes RADVISION's leadership in video quality assessment based on 17 years of experience in video conferencing.
Cisco Connect 2018 Thailand - Investment protection; insure your return on co...NetworkCollaborators
This document discusses optimizing Cisco collaboration deployments and operations through continuous performance monitoring and issue resolution. It recommends assessing the current network, identifying and resolving issues, and continuously monitoring to address problems. This approach can improve user experience, accelerate deployments, increase operational efficiency, and optimize resource utilization over the Cisco solution lifecycle. IR partners with Cisco to provide the Prognosis communication performance management solution, which has been tested and certified for Cisco environments.
Automatically Generated Simulations for Predicting Software-Defined Networkin...Felipe Alencar
This document proposes a new modeling language called Network Modeling Language (NML) to model software-defined networking (SDN) and automatically generate simulations using different performance models like queueing Petri nets (QPN) and stochastic Petri nets (SPN). It presents the NML metamodel and model transformations from NML to QPN and SPN. Experiments comparing the accuracy and simulation time of the generated QPN and SPN models against real measurements on a SDN testbed show that QPN predictions have 3% error while SPN simulations are faster but less accurate. The work contributes an approach for automatic performance prediction of SDN networks from a single high-level model.
This document provides an overview of speech quality testing solutions. It discusses the development of subjective and objective speech quality testing methods. PESQ and POLQA are described as the two main objective testing algorithms. PESQ became an ITU standard in 2001 and provides accurate quality predictions. POLQA was developed later to support new codecs, super-wideband speech, and handle factors like time variation. It will become the new recommended ITU standard. The document also introduces Dingli's speech quality testing solutions which are based on products like Pilot Pioneer and use PESQ and POLQA algorithms.
The document discusses the effectiveness and efficiency of different types of network test solutions for identifying issues in video delivery. Stateful traffic testing that emulates real applications from layer 2-7 is more effective than stateless testing as it provides more accurate performance data and prediction of network issues. Using real sample traffic from multiple devices and formats allows more accurate modeling and understanding of network behavior under different conditions. Testing needs to evolve to address new video applications and protocols like IPv6.
The presentation is ideal for people who needs to know how Bluecoat packetshaper could be beneficial to any organization. Especially with organizations having problems with Data network visibility traversing across WAN/LAN traffic.
Improved voice quality with the combination of transport layer & audio codec ...journalBEEI
Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec & signaling protocol produced the quality speech.
Video quality measurements can be performed using subjective, objective, and payload-based methods. Subjective methods involve human assessment while objective methods use measurement devices and are repeatable for testing and monitoring. Payload-based methods assess video quality by comparing the original and distorted video. Standardization bodies have defined various levels of measurement including transport, transaction, and content levels to analyze video quality from different perspectives.
Comparisons of QoS in VoIP over WIMAX by Varying the Voice codes and Buffer sizeEditor IJCATR
Voice over Internet Protocol (VoIP) is developed for voice communications system based on voice packets transmitted over
IP network with real-time communications of voice across networks using the Internet protocols. Quality of Service (QoS) mechanism
is applied to guarantee successful voice packets transmitted over IP network with reduced delay or drop according to assigned priority
of voice packets. In this paper, the goal of simulation models is present to investigate the performance of VoIP codecs and buffer size
for improving quality of service (QoS) with the simulation results by using OPNET modeler version 14.5. The performance of the
proposed algorithm is analyzed and compared the quality of service for VoIP. The final simulated result shows that the VoIP service
performance best under G.729 voice encoder scheme and buffer size 256 Kb over WiMAX network.
The document summarizes a PhD thesis defense about novel applications for emerging markets using television as a ubiquitous device. The thesis proposed improving the user experience of TV as a low-cost internet access device through:
1) QoS-aware video transmissions and low-complexity video security for applications like video chat and distance education.
2) Context-aware intelligent TV-internet mashups using TV channel identity and text recognition from static and broadcast video as context.
3) A novel on-screen keyboard for text entry using a TV remote for non-computer savvy users. User studies on early prototypes identified challenges with interfaces, video quality, and text entry that the thesis aimed to address.
Text independent speaker recognition systemDeepesh Lekhak
This document outlines a project to develop a text-independent speaker recognition system. It lists the project members and provides an overview of the presentation sections, which include the system architecture, methodology, results and analysis, and applications. The methodology section describes implementing the system in MATLAB, including voice capturing, pre-processing, MFCC feature extraction, GMM matching, and identification/verification. It also outlines implementing the system on an FPGA, including analog conversion, storage, framing, FFT, mel spectrum, MFCC extraction, and UART transmission to MATLAB for further processing. The results show over 99% recognition accuracy with longer training and test data.
VoIP Monitoring and Analysis - Still Top of Mind in Network Performance Monit...Savvius, Inc
With over 10 years of deployment history, VoIP is the primary voice solution for just about every company in existence - large, medium, or small. But even with all that history, recent research from TRAC shows that VoIP is still the number one IT initiative impacting network performance. And with the growth of 802.11 and Wi-Fi enabled smart phones, the use of voice over Wi-Fi (VoFi) promises to increase the volume of VoIP traffic even more. Analyzing VoIP traffic alone is not enough. VoIP analysis must be part of your overall network performance analysis. After all, VoIP is just another data type on your network, and according to TRAC, it is impacting your network performance, so you must monitor and analyze the network as a whole, including voice and video over IP. Join us to see how easy it is to capture and analyze voice, video, and data traffic simultaneously, allowing you to pinpoint the impact of each data type on your overall network performance.
The document summarizes the current state of research on Voice over IP (VoIP) quality of service (QoS) in wireless mesh networks (WMNs). It provides background on the challenges of supporting VoIP in dynamic WMNs. The author's research plan involves evaluating VoIP performance under different QoS feature modifications, including adding fixed mesh routers and using IPv6. Simulations will measure delay, jitter and packet loss under scenarios with varying mobility and traffic conditions. Progress so far includes a published paper and near completion of a second paper for submission.
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This is inspired from Tom Mitchell's book on Machine Learning. You can achieve a bit exact implementation of the back propagation algorithm if you follow the code in this.
This document outlines the requirements for a job searching software system. It will collect information from companies and candidates, evaluate their requirements, and match candidates to job openings. Companies will provide details like job title, location, and qualifications required. Candidates will provide personal information, education history, preferred locations and industries. The software will score and filter candidates for each job based on how well their qualifications match the company's requirements. Candidates that meet all required criteria will be presented as potential matches to help both job seekers and employers find suitable opportunities.
NUAV - A Testbed for Development of Autonomous Unmanned Aerial Vehiclesadil raja
The document proposes a testbed called NUAV for developing autonomous unmanned aerial vehicles (UAVs). NUAV integrates the flight simulator FlightGear with machine learning algorithms. It allows simulating UAVs for various scenarios. A simple experiment uses a neural network optimized by a genetic algorithm to fly a simulated aircraft. The testbed aims to enable training UAVs to perform user-defined tasks and developing fleets of cooperative autonomous UAVs.
The document discusses DevOps concepts including tools, development processes, code scripting, and conclusions. It provides an outline and covers topics like continuous delivery, Docker containers, programming in Java and Groovy, and errors in flight control code. Code examples are provided to illustrate programming concepts and a Groovy script is shown.
The document discusses research and development at an institution. It outlines topics like research proposals, basic research themes, issues in research, and cloud computing. Specifically, it proposes deploying a private cloud to address computing needs for research through open-source software like OpenStack. However, there are challenges to implementing a private cloud like hardware costs, space, electricity, internet access, and manpower constraints that would need to be addressed.
This document outlines a server and client for remote command execution. The server runs on a specified port, accepts connections from clients, and executes commands sent by the client using a bash process. It returns the output to the client. The client connects to the server, takes input from the user and sends it to the server, and prints the response until it receives a "bye" message. References are provided for the source code and presentation platform used.
This document is a thesis submitted by Muhammad Adil Raja to the Department of Computer Science at Lahore University of Management Sciences in partial fulfillment of the requirements for a Masters of Science degree in Computer Science. The thesis focuses on implementing and evaluating a routing algorithm for sensor networks based on minimum transmission energy routing. It includes chapters on sensor networks and routing protocols, describing the proposed algorithm and simulator for evaluating it, and analyzing the results of testing the algorithm on different network topologies.
This document discusses data mining techniques, including decision trees. It describes the basic steps in data mining as exploration, model building and validation, and deployment. It then discusses some common techniques used in data mining like association analysis, decision trees, neural networks, and statistical methods. It focuses on decision trees, describing how they take an input object and output a yes/no decision. Decision trees can represent both classification and regression problems depending on whether the target variable is categorical or continuous. The document discusses how decision trees examine predictor variables one at a time to determine the best splits to minimize misclassification.
This document defines VoIP and its protocols. VoIP allows routing of phone calls over IP networks using packet switching instead of traditional circuit switching. Key protocols include H.323 for call setup, control and transport, and RTP for real-time media transport over UDP. H.323 defines codecs for digitizing and compressing voice, and uses signaling protocols like H.225 and H.245 as well as gatekeepers for call routing and quality of service control. RTCP monitors RTP transport quality. H.323 gateways enable interworking between IP and circuit-switched networks.
The document discusses the specifications and communication methodology for the ULMAN GUI. The GUI is required to implement communications and control functions, network monitoring, log file functions, and statistics analysis. It communicates securely with ULMAN wireless nodes using SSH and SCP for file transfers. Each node runs an SSH daemon while the super node runs an SSH client as a backend service for the GUI.
Modeling the Effect of packet Loss on Speech Quality: GP Based Symbolic Regre...adil raja
This document describes research using genetic programming to model the effect of packet loss on speech quality. It examines previous approaches using packet-based models, speech quality models, and neural networks. The current research uses genetic programming and a Gilbert model of packet loss to simulate loss and develop a model relating loss metrics to mean opinion scores. The model is trained and validated on speech files with varying packet loss, and genetic programming is used to develop functions mapping loss metrics to quality scores.
Modelling the Effect of Packet Loss on Speech Qualityadil raja
The document proposes a new method for modeling the effect of packet loss on speech quality. It summarizes existing packet-based and speech-based approaches before describing a neural network model that takes packet loss parameters like mean loss rate, burst length, and gap length as inputs. The proposed method achieves a correlation of 0.9807 compared to 0.952 and 0.93 for previous methods. Scatter plots and an error plot are provided to analyze the model's performance.
The document discusses Muhammad Adil Raja's research project on non-intrusive real-time quality assessment of speech for VoIP using Hidden Markov Models. It describes using HMMs with a mixture density function and a few Gaussians to calculate the probability of an observation sequence and find the most likely state sequence that could have emitted it, based on the Viterbi algorithm and dynamic programming. It asks for recommendations on applications that can access the Viterbi probability, mentioning HTK, HMM Toolbox by Kevin Murphy, and Sphinx4 as options already considered.
This document provides an overview of the IXA SDK 3.5 simulation environment and programming models. It discusses how the SDK provides packet simulation options and code for basic router functions like receive, process, and transmit. Specifically, it presents code snippets for a static forwarding application built from microblocks and core components that are executed across multiple microengines and an XScale core.
The document summarizes research on wireless access and packet processing. It describes an application that uses two ports to bridge and process received packets. Packets are received, moved to memory for processing by microengines, and transmitted. Testing showed the application achieved 100% throughput for packet sizes from 64 to 1518 bytes. Latency increased with packet size and load, while no packet loss occurred.
The document summarizes research on wireless access and packet processing. It describes an application that uses two ports to bridge and process received packets. Packets are received, moved to memory for processing, and transmitted. Processing includes calculating metrics like delay and jitter. Tests show the application achieved 100% throughput for packet sizes from 64 to 1518 bytes and experienced no packet loss. Latency increased with packet size and load.
Comparative analysis between traditional aquaponics and reconstructed aquapon...bijceesjournal
The aquaponic system of planting is a method that does not require soil usage. It is a method that only needs water, fish, lava rocks (a substitute for soil), and plants. Aquaponic systems are sustainable and environmentally friendly. Its use not only helps to plant in small spaces but also helps reduce artificial chemical use and minimizes excess water use, as aquaponics consumes 90% less water than soil-based gardening. The study applied a descriptive and experimental design to assess and compare conventional and reconstructed aquaponic methods for reproducing tomatoes. The researchers created an observation checklist to determine the significant factors of the study. The study aims to determine the significant difference between traditional aquaponics and reconstructed aquaponics systems propagating tomatoes in terms of height, weight, girth, and number of fruits. The reconstructed aquaponics system’s higher growth yield results in a much more nourished crop than the traditional aquaponics system. It is superior in its number of fruits, height, weight, and girth measurement. Moreover, the reconstructed aquaponics system is proven to eliminate all the hindrances present in the traditional aquaponics system, which are overcrowding of fish, algae growth, pest problems, contaminated water, and dead fish.
Discover the latest insights on Data Driven Maintenance with our comprehensive webinar presentation. Learn about traditional maintenance challenges, the right approach to utilizing data, and the benefits of adopting a Data Driven Maintenance strategy. Explore real-world examples, industry best practices, and innovative solutions like FMECA and the D3M model. This presentation, led by expert Jules Oudmans, is essential for asset owners looking to optimize their maintenance processes and leverage digital technologies for improved efficiency and performance. Download now to stay ahead in the evolving maintenance landscape.
Rainfall intensity duration frequency curve statistical analysis and modeling...bijceesjournal
Using data from 41 years in Patna’ India’ the study’s goal is to analyze the trends of how often it rains on a weekly, seasonal, and annual basis (1981−2020). First, utilizing the intensity-duration-frequency (IDF) curve and the relationship by statistically analyzing rainfall’ the historical rainfall data set for Patna’ India’ during a 41 year period (1981−2020), was evaluated for its quality. Changes in the hydrologic cycle as a result of increased greenhouse gas emissions are expected to induce variations in the intensity, length, and frequency of precipitation events. One strategy to lessen vulnerability is to quantify probable changes and adapt to them. Techniques such as log-normal, normal, and Gumbel are used (EV-I). Distributions were created with durations of 1, 2, 3, 6, and 24 h and return times of 2, 5, 10, 25, and 100 years. There were also mathematical correlations discovered between rainfall and recurrence interval.
Findings: Based on findings, the Gumbel approach produced the highest intensity values, whereas the other approaches produced values that were close to each other. The data indicates that 461.9 mm of rain fell during the monsoon season’s 301st week. However, it was found that the 29th week had the greatest average rainfall, 92.6 mm. With 952.6 mm on average, the monsoon season saw the highest rainfall. Calculations revealed that the yearly rainfall averaged 1171.1 mm. Using Weibull’s method, the study was subsequently expanded to examine rainfall distribution at different recurrence intervals of 2, 5, 10, and 25 years. Rainfall and recurrence interval mathematical correlations were also developed. Further regression analysis revealed that short wave irrigation, wind direction, wind speed, pressure, relative humidity, and temperature all had a substantial influence on rainfall.
Originality and value: The results of the rainfall IDF curves can provide useful information to policymakers in making appropriate decisions in managing and minimizing floods in the study area.
Introduction- e - waste – definition - sources of e-waste– hazardous substances in e-waste - effects of e-waste on environment and human health- need for e-waste management– e-waste handling rules - waste minimization techniques for managing e-waste – recycling of e-waste - disposal treatment methods of e- waste – mechanism of extraction of precious metal from leaching solution-global Scenario of E-waste – E-waste in India- case studies.
artificial intelligence and data science contents.pptxGauravCar
What is artificial intelligence? Artificial intelligence is the ability of a computer or computer-controlled robot to perform tasks that are commonly associated with the intellectual processes characteristic of humans, such as the ability to reason.
› ...
Artificial intelligence (AI) | Definitio
Electric vehicle and photovoltaic advanced roles in enhancing the financial p...IJECEIAES
Climate change's impact on the planet forced the United Nations and governments to promote green energies and electric transportation. The deployments of photovoltaic (PV) and electric vehicle (EV) systems gained stronger momentum due to their numerous advantages over fossil fuel types. The advantages go beyond sustainability to reach financial support and stability. The work in this paper introduces the hybrid system between PV and EV to support industrial and commercial plants. This paper covers the theoretical framework of the proposed hybrid system including the required equation to complete the cost analysis when PV and EV are present. In addition, the proposed design diagram which sets the priorities and requirements of the system is presented. The proposed approach allows setup to advance their power stability, especially during power outages. The presented information supports researchers and plant owners to complete the necessary analysis while promoting the deployment of clean energy. The result of a case study that represents a dairy milk farmer supports the theoretical works and highlights its advanced benefits to existing plants. The short return on investment of the proposed approach supports the paper's novelty approach for the sustainable electrical system. In addition, the proposed system allows for an isolated power setup without the need for a transmission line which enhances the safety of the electrical network
Null Bangalore | Pentesters Approach to AWS IAMDivyanshu
#Abstract:
- Learn more about the real-world methods for auditing AWS IAM (Identity and Access Management) as a pentester. So let us proceed with a brief discussion of IAM as well as some typical misconfigurations and their potential exploits in order to reinforce the understanding of IAM security best practices.
- Gain actionable insights into AWS IAM policies and roles, using hands on approach.
#Prerequisites:
- Basic understanding of AWS services and architecture
- Familiarity with cloud security concepts
- Experience using the AWS Management Console or AWS CLI.
- For hands on lab create account on [killercoda.com](https://killercoda.com/cloudsecurity-scenario/)
# Scenario Covered:
- Basics of IAM in AWS
- Implementing IAM Policies with Least Privilege to Manage S3 Bucket
- Objective: Create an S3 bucket with least privilege IAM policy and validate access.
- Steps:
- Create S3 bucket.
- Attach least privilege policy to IAM user.
- Validate access.
- Exploiting IAM PassRole Misconfiguration
-Allows a user to pass a specific IAM role to an AWS service (ec2), typically used for service access delegation. Then exploit PassRole Misconfiguration granting unauthorized access to sensitive resources.
- Objective: Demonstrate how a PassRole misconfiguration can grant unauthorized access.
- Steps:
- Allow user to pass IAM role to EC2.
- Exploit misconfiguration for unauthorized access.
- Access sensitive resources.
- Exploiting IAM AssumeRole Misconfiguration with Overly Permissive Role
- An overly permissive IAM role configuration can lead to privilege escalation by creating a role with administrative privileges and allow a user to assume this role.
- Objective: Show how overly permissive IAM roles can lead to privilege escalation.
- Steps:
- Create role with administrative privileges.
- Allow user to assume the role.
- Perform administrative actions.
- Differentiation between PassRole vs AssumeRole
Try at [killercoda.com](https://killercoda.com/cloudsecurity-scenario/)
Software Engineering and Project Management - Introduction, Modeling Concepts...Prakhyath Rai
Introduction, Modeling Concepts and Class Modeling: What is Object orientation? What is OO development? OO Themes; Evidence for usefulness of OO development; OO modeling history. Modeling
as Design technique: Modeling, abstraction, The Three models. Class Modeling: Object and Class Concept, Link and associations concepts, Generalization and Inheritance, A sample class model, Navigation of class models, and UML diagrams
Building the Analysis Models: Requirement Analysis, Analysis Model Approaches, Data modeling Concepts, Object Oriented Analysis, Scenario-Based Modeling, Flow-Oriented Modeling, class Based Modeling, Creating a Behavioral Model.
Realtime, Non-Intrusive Evaluation of VoIP Using Genetic Programming
1. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
REAL-TIME, NON-INTRUSIVE EVALUATION OF
VOIP
USING GENETIC PROGRAMMING
A. Raja1 A. Azad2 C. Flanagan1 C. Ryan2
1Wireless Access Research Centre
Department of Electronic and Computer Engineering
2Bio-Computing and Developmental Systems
Department of Computer Science and Information Sysmtems
University of Limerick, Limerick, Ireland
EuroGP 2007 – 10th European conference on Genetic
Programming
2. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OUTLINE
1 MOTIVATION
Preamble
The Problem of Speech Quality Assessment
Voice Over IP
Research Goal
2 VOIP SIMULATION ENVIRONMENT
Simulation System
Network Traffic Characteristics
3 GP EXPERIMENTS
4 TEST RESULTS
5 CONCLUSIONS
3. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OUTLINE
1 MOTIVATION
Preamble
The Problem of Speech Quality Assessment
Voice Over IP
Research Goal
2 VOIP SIMULATION ENVIRONMENT
Simulation System
Network Traffic Characteristics
3 GP EXPERIMENTS
4 TEST RESULTS
5 CONCLUSIONS
4. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OUTLINE
1 MOTIVATION
Preamble
The Problem of Speech Quality Assessment
Voice Over IP
Research Goal
2 VOIP SIMULATION ENVIRONMENT
Simulation System
Network Traffic Characteristics
3 GP EXPERIMENTS
4 TEST RESULTS
5 CONCLUSIONS
5. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OUTLINE
1 MOTIVATION
Preamble
The Problem of Speech Quality Assessment
Voice Over IP
Research Goal
2 VOIP SIMULATION ENVIRONMENT
Simulation System
Network Traffic Characteristics
3 GP EXPERIMENTS
4 TEST RESULTS
5 CONCLUSIONS
6. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OUTLINE
1 MOTIVATION
Preamble
The Problem of Speech Quality Assessment
Voice Over IP
Research Goal
2 VOIP SIMULATION ENVIRONMENT
Simulation System
Network Traffic Characteristics
3 GP EXPERIMENTS
4 TEST RESULTS
5 CONCLUSIONS
7. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
PREAMBLE
VoIP – A paradigm shift
Bandwidth redundancy exploitation
QoS remains dominated by network/transport layer
degradations
Quality assessment ...
Reflects upon the operating conditions of the network
8. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
PREAMBLE
VoIP – A paradigm shift
Bandwidth redundancy exploitation
QoS remains dominated by network/transport layer
degradations
Quality assessment ...
Reflects upon the operating conditions of the network
9. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
PREAMBLE
VoIP – A paradigm shift
Bandwidth redundancy exploitation
QoS remains dominated by network/transport layer
degradations
Quality assessment ...
Reflects upon the operating conditions of the network
10. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
PREAMBLE
VoIP – A paradigm shift
Bandwidth redundancy exploitation
QoS remains dominated by network/transport layer
degradations
Quality assessment ...
Reflects upon the operating conditions of the network
11. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
PREAMBLE
VoIP – A paradigm shift
Bandwidth redundancy exploitation
QoS remains dominated by network/transport layer
degradations
Quality assessment ...
Reflects upon the operating conditions of the network
12. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
PREAMBLE
VoIP – A paradigm shift
Bandwidth redundancy exploitation
QoS remains dominated by network/transport layer
degradations
Quality assessment ...
Reflects upon the operating conditions of the network
13. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
SPEECH QUALITY ASSESSMENT METHODOLOGIES
Two approaches to speech quality Assessment
1 Subjective Assessment
2 Objective Assessment
14. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
SUBJECTIVE ASSESSMENT OF SPEECH QUALITY
Speech quality is estimated by humans.
Advantage – Reliable results.
Limitations
1 Expensive
2 Time Consuming
3 Laborious
4 Lack of Repeatability
Mean Opinion Score (MOS) is the measure of quality.
1 – bad
5 – Excellent
15. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
A computer automated fast and reliable program is used to
assay human perception of speech quality
Two approaches:
1 Intrusive Assessment
2 Non-Intrusive Assessment
16. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
INTRUSIVE ASSESSMENT
The signal under test is compared against a corresponding
reference signal.
Advantages:
1 The most reliable artificial means of estimating speech
quality
2 Tests can be repeated easily
Limitations:
1 Consumes considerable computing resources.
2 Is not useful for continuous monitoring of quality due to
requirement of a reference signal.
17. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
INTRUSIVE ASSESSMENT
The signal under test is compared against a corresponding
reference signal.
Advantages:
1 The most reliable artificial means of estimating speech
quality
2 Tests can be repeated easily
Limitations:
1 Consumes considerable computing resources.
2 Is not useful for continuous monitoring of quality due to
requirement of a reference signal.
18. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
ITU-T P.862 (PESQ)
PESQ algorithm is the current ITU-T Recommendation for
intrusive speech quality estimation.
The speech signal is mapped from time domain to
time-frequency representation using the psychophysical
equivalents of frequency and intensity.
19. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
ITU-T P.862 (PESQ)
It has shown a high correlation with various ITU-T
benchmark tests.
For 30 ITU-T subjective tests the Pearson’s Correlation
Coefficient (R) was 0.935
20. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
NON-INTRUSIVE ASSESSMENT
A challenging problem since a reference is not available.
Two approaches exist
1 Signal-based models
2 Parametric models
Signal-based models
Recent approaches are based on emulating
1 Human speech production model
2 Psychoacoustic processing of human ear
ITU-T P.563 is the current Recommendation.
21. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
NON-INTRUSIVE ASSESSMENT
A challenging problem since a reference is not available.
Two approaches exist
1 Signal-based models
2 Parametric models
Signal-based models
Recent approaches are based on emulating
1 Human speech production model
2 Psychoacoustic processing of human ear
ITU-T P.563 is the current Recommendation.
22. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
OBJECTIVE ASSESSMENT OF SPEECH QUALITY
PARAMETRIC MEASUREMENT OF VOIP QUALITY
Functions of transport layer metrics and other measurable
quantities.
Cogent metrics may be:
Packet Loss Rate
Variable delay – jitter
End-to-end delay
. . .
Aimed at Real-time and continuous evaluation of quality
23. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
VOICE OVER IP – VOIP
Packet based communication channel
Uses wire-line speech codecs
Linear Predictive Coding (LPC) is having vogue
Coded frames are packetized into RTP/UDP
Internet is used for transportation
The receiver does the reverse process
24. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
RESEARCH GOAL
Derivation of a VoIP listening Quality estimation model as a
function of transport layer metrics.
Genetic Programming based Symbolic Regression is used
Using the PESQ algorithm as the reference system
25. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
VOIP SIMULATION ENVIRONMENT
PACKET LOSS SIMULATION – THE GILBERT ELLIOT MODEL
mlr = p
p+q (1)
mbl = 1
q (2)
clp = 1 − q (3)
mbl = 1
1−clp (4)
Where
mlr – mean loss rate
mbl – mean burst length
clp – conditional loss probability
26. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
VOIP SIMULATION ENVIRONMENT
27. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
NETWORK TRAFFIC PARAMETERS
No. Parameter Name Abbreviation
1 Bit-rate (kbps) br
2 mean loss rate mlr
3 mean burst length mbl
4 Packetization Interval (ms) PI
5 Frame duration (ms) fd
28. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
NETWORK TRAFFIC SCENARIOS
No. Parameter Range
1 br G.729 (8 kbps), G.723.1 (6.3 kbps),
AMR 7.4 and 12.2 kbps
2 mlr [0,2.5,3.5,. . . 15,20,25,. . . 40]%
3 mbl 10, 50, 60, 70 and 80%
4 PI 10-60 ms
5 fd 10, 20, 30 ms
29. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
EXPERIMENTAL SETUP
GPLab
Four GP Experiments were performed with various
configurations
Commonalities
Each experiment constituted 50 runs
Each Run spanned 50 generations
30. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
GP EXPERIMENTS
COMMON PARAMETERS
Parameter Value
Initial Population Size 300
Selection LPP Tournament
Tournament Size 2
Genetic Operators Crossover and Subtree Mutation
Initial Operator probabilities 0.5 initial, adaptive onwards
Survival Half Elitism
Function Set +, -, *, /, sin, cos, log2, log10,
loge, sqrt, power,
Terminal Set Random numbers [0.0 . . . 1.0]
Integers [2 . . . 10]. mlrVAD,
mblVAD, PI, br, fd
31. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
GP EXPERIMENTS
EXPERIMENTAL DETAILS
Experiment 1:
Fitness function – Mean Squared Error MSE
Experiment 2:
Linear Scaling MSEs
MSEs(y, t) = 1/n
n
i
(ti − (a + byi))2
(5)
a = t − by, b =
cov(t, y)
var(y)
(6)
32. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
GP EXPERIMENTS
EXPERIMENTAL DETAILS
Experiment 1:
Fitness function – Mean Squared Error MSE
Experiment 2:
Linear Scaling MSEs
MSEs(y, t) = 1/n
n
i
(ti − (a + byi))2
(5)
a = t − by, b =
cov(t, y)
var(y)
(6)
33. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
GP EXPERIMENTS
EXPERIMENTAL DETAILS
Experiments 3 and 4
Selection criterion based on Gustafson et al. was used
Mating takes place between dissimilar individuals
Experiment 4:
The Maximum tree depth was reduced to 7 from 17
The results were treated to Mann-Whitney-Wilcoxon Test
for significance Analysis
Experiment 4 was found to be significantly better overall.
34. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
GP EXPERIMENTS
EXPERIMENTAL DETAILS
Experiments 3 and 4
Selection criterion based on Gustafson et al. was used
Mating takes place between dissimilar individuals
Experiment 4:
The Maximum tree depth was reduced to 7 from 17
The results were treated to Mann-Whitney-Wilcoxon Test
for significance Analysis
Experiment 4 was found to be significantly better overall.
35. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
ON DATA COLLECTION
Nortel ND speech database containing high quality signals
with speech from 2 male and 2 female speakers was used
for analysis.
A total of 3360 distorted speech files were created for each
combination of network traffic parameters.
1177 35% were used for training
503 15% were used for testing
1680 50% were used for speaker independent validation
36. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
VOIP QUALITY MONITORING MODELS
MOS − LQOGP = −2.46 × log(cos(log(br)) + mlrVAD
×(br + fd/10)) + 3.17 (7)
MOS − LQOGP = −2.99 × cos(0.91 × sin(mlrVAD)
+mlrVAD + 8) + 4.20 (8)
Equation (7) Equation(8)
Data MSEs σ MSEs σ
Training 0.0370 0.9634 0.0520 0.9481
Testing 0.0387 0.9646 0.0541 0.9501
Validation 0.0382 0.9688 0.0541 0.9531
38. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
SCATTER PLOTS
ON PERFORMANCE OF ITU-T P.563
39. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
CONCLUSIONS
1 The model is a good approximation to PESQ.
2 Suitable for real-time and non-intrusive estimation of
speech quality whereas PESQ is NOT.
3 Simple models; depend on 3 and 1 variable respectively.
4 Performs significantly better than ITU-T P.563
40. Motivation VoIP Simulation Environment GP Experiments Test Results Conclusions
FUTURE GOALS
To include wide band codecs in the research.
To develop a unified quality estimation model for narrow
and wide band telephony