Benefits of easy integration of a new voice quality measurement software providing proof of concenpt and test results. Describes how easy integration process can be, advantages comparing with PESQ providing explicit test results.
AQuA Wiki - enabling voice quality monitoring in AsteriskSevana Oü
AQuA – Audio Quality Analyzer, a product of Sevana Oy, Finland is a simple but powerful tool to provide perceptual voice quality testing and audio quality monitoring. This is the easiest way to compare two audio files and test voice quality between original and degraded files. Technology implemented in AQuA is based on the research of World recognized scientists and allows receiving metrics of perceptual audio quality measurement for voice, HD Voice and wideband audio signals.
Niqa competitive alternative for non-intrusive voice quality testing (p.563)Sevana Oü
The document discusses non-intrusive voice quality testing using NIQA as a competitive alternative to P.563, noting the limitations of traditional approaches and how NIQA uses psychoacoustic algorithms and trained associations to more accurately assess voice quality and identify reasons for degradation, with applications including audio conferencing, VoIP monitoring, and network optimization.
Automated Voice And Audio Quality Test Measurementguest5a90cfc
AQuA is a simple but powerful tool to provide perceptual voice quality testing and audio files comparison. This is the easiest way to compare two sound signals and test voice quality between original and degraded files.
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.
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.
Interest towards speech coding & standardization:
– World wide growth in communication networks
– Emergence of new multimedia applications
– Advances in Very Large-Scale Integration (VLSI)
devices
• Standardization
– International Telecommunications Union (ITU)
– European Telecom. Standards Institute (ETSI)
– International Standards Organization (ISO)
– Telecommunication Industry Association (TIA), NA
– R&D Center for Radio systems (RCR), Japan
- Adaptive Multi-Rate (AMR) speech coding allows variable bit rates depending on channel conditions between 4.75 kbit/s and 12.2 kbit/s over full-rate channels and between 4.75 kbit/s and 7.95 kbit/s over half-rate channels. It uses source coding, channel coding, and rate adaptation based on channel estimation to optimize quality and efficiency.
- AMR utilizes algebraic code excited linear prediction speech coding, unequal error protection, recursive systematic convolutional channel coding, and discontinuous transmission with voice activity detection and comfort noise generation. This allows it to save bandwidth during silence and adapt to changing channel conditions.
1998년 Soundcheck 이란 제품이 미국내 오디오 측정 분 Best Application 상을 받게 되었읍니다. 현 전세계 많은 사용자가 Soundcheck 을 사용하여 오디오 분석을 하고 있으며 많은 Loudspeaker제조회사 , 보청기 제조회사 , 무선 마이크 및 블루투스 이어폰 , 헤드셋 제조 회사들이 사용하고 있읍니다.
AQuA Wiki - enabling voice quality monitoring in AsteriskSevana Oü
AQuA – Audio Quality Analyzer, a product of Sevana Oy, Finland is a simple but powerful tool to provide perceptual voice quality testing and audio quality monitoring. This is the easiest way to compare two audio files and test voice quality between original and degraded files. Technology implemented in AQuA is based on the research of World recognized scientists and allows receiving metrics of perceptual audio quality measurement for voice, HD Voice and wideband audio signals.
Niqa competitive alternative for non-intrusive voice quality testing (p.563)Sevana Oü
The document discusses non-intrusive voice quality testing using NIQA as a competitive alternative to P.563, noting the limitations of traditional approaches and how NIQA uses psychoacoustic algorithms and trained associations to more accurately assess voice quality and identify reasons for degradation, with applications including audio conferencing, VoIP monitoring, and network optimization.
Automated Voice And Audio Quality Test Measurementguest5a90cfc
AQuA is a simple but powerful tool to provide perceptual voice quality testing and audio files comparison. This is the easiest way to compare two sound signals and test voice quality between original and degraded files.
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.
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.
Interest towards speech coding & standardization:
– World wide growth in communication networks
– Emergence of new multimedia applications
– Advances in Very Large-Scale Integration (VLSI)
devices
• Standardization
– International Telecommunications Union (ITU)
– European Telecom. Standards Institute (ETSI)
– International Standards Organization (ISO)
– Telecommunication Industry Association (TIA), NA
– R&D Center for Radio systems (RCR), Japan
- Adaptive Multi-Rate (AMR) speech coding allows variable bit rates depending on channel conditions between 4.75 kbit/s and 12.2 kbit/s over full-rate channels and between 4.75 kbit/s and 7.95 kbit/s over half-rate channels. It uses source coding, channel coding, and rate adaptation based on channel estimation to optimize quality and efficiency.
- AMR utilizes algebraic code excited linear prediction speech coding, unequal error protection, recursive systematic convolutional channel coding, and discontinuous transmission with voice activity detection and comfort noise generation. This allows it to save bandwidth during silence and adapt to changing channel conditions.
1998년 Soundcheck 이란 제품이 미국내 오디오 측정 분 Best Application 상을 받게 되었읍니다. 현 전세계 많은 사용자가 Soundcheck 을 사용하여 오디오 분석을 하고 있으며 많은 Loudspeaker제조회사 , 보청기 제조회사 , 무선 마이크 및 블루투스 이어폰 , 헤드셋 제조 회사들이 사용하고 있읍니다.
Course Content of 6ET3 Digital communication,of B.E. VIth Semester (Electronics & Telecommunication Engineering) of Sant Gadge Baba Amravati University, Maharashtra, India
Speech coding is used to efficiently transmit speech through digital channels by retaining only the information useful to listeners. The LPC-10 standard uses linear predictive coding with 10 coefficients to analyze and synthesize speech. During analysis, it extracts parameters like voicing and pitch from the speech signal. The synthesis process uses these parameters to generate noise or periodic excitation, apply an LPC filter, and control gain. LPC-10 transmits speech at 2.4kbps by coding the 10 LPC coefficients, pitch, voicing, and energy into 54 bits per frame. It enables understandable but unnatural sounding speech and is used for secure voice transmissions.
This document discusses various techniques for speech coding used in digital communication systems. It covers fundamental concepts like sampling theory, quantization, predictive coding, and linear predictive coding (LPC). It then describes specific speech codecs including PCM, ADPCM, CELP, LD-CELP, ACELP, and LPC vocoders. It discusses characteristics of speech coding like being lossy and metrics like SNR and MOS. Finally, it provides details on widely used standards like G.711, G.729, G.723.1, and GSM.
This document provides an overview of random processes, which are used to model information sources in source coding and communications systems. It first defines basic probability concepts like events, sample spaces, and the axioms of probability. It then introduces random variables as mappings from a sample space to real numbers, and random processes as functions of a random variable and time. Common random process models are described, including memoryless, Markov, and stationary processes. Understanding random processes allows analyzing source coding performance based on probabilistic source models.
This document discusses parameters for measuring speech quality in cellular networks. It describes how speech quality is an important aspect of mobile communication quality. Key parameters for measuring speech quality include the Speech Quality Index (SQI), which estimates speech quality as perceived by a human listener. Other parameters discussed include RxQual, Bit Error Rate, Frame Error Rate, and L3 messages. The document also discusses intrusive and non-intrusive methods for measuring speech quality, as well as scales like Mean Opinion Score and Perceptual Evaluation of Speech Quality. Network performance is evaluated using these speech quality measurements and other key performance indicators.
Multirate signal processing and decimation interpolationransherraj
This document is a report on multirate signal processing, decimation, and interpolation. It begins with an introduction that defines multirate signal processing as using signals with different sampling frequencies. It then discusses decimation, which decreases the sampling rate by removing samples, and interpolation, which increases the sampling rate by estimating values between known samples. Applications of multirate signal processing are also discussed, such as digital audio and speech processing. The report concludes that changing the sampling frequency through decimation and interpolation can increase processing efficiency.
The document discusses multi-dimensional approaches to measuring voice quality over packet networks. It describes measuring three key dimensions: delay, echo, and clarity. Delay is one of the most challenging aspects, as real-time voice requires low delay. The document outlines various sources of delay in voice over packet (VoP) networks and techniques for measuring round-trip and one-way delay using specialized instruments and digital signal processing algorithms.
The document discusses GSM network optimization techniques including adjusting parameters for cell selection, power control, and handover control to improve coverage, interference, and handover behavior. It describes the optimization process including initial, primary, and maintenance phases. Key parameters and techniques discussed include enabling features like discontinuous transmission, frequency hopping, and power control as well as adjusting neighbor cell lists, antenna configuration, and frequencies. Drive testing is used to identify problems, verify solutions, and ensure quality.
The document compares the performance of single stage and double stage interleavers in communication systems using turbo codes. A single stage interleaver uses one random interleaver between two convolutional encoders, while a double stage interleaver uses two interleavers in series. The document suggests that a double stage interleaver can improve the bit error rate (BER) of the system compared to a single stage interleaver by further scrambling the input bits. It also provides details on the components of a turbo code system such as convolutional encoders, interleavers, puncturing, and iterative decoding.
This document describes Fracton Technologies and its network optimization solutions. Fracton is an Indian company that focuses on innovations in radio access network optimization for GSM, UMTS, and LTE technologies. It offers a range of network optimization services and has an outstanding track record of optimizing networks for leading mobile operators. Its flagship product is MaxCell, an automated optimization tool that optimizes network performance and quality of experience by tuning cell database parameters based on network configuration and performance data.
IRJET- Segmentation in Digital Signal ProcessingIRJET Journal
1) The document discusses audio signal segmentation techniques in digital signal processing. It analyzes algorithms for segmenting audio signals into homogeneous segments for applications like audio event recognition and speaker identification.
2) A hybrid classification approach is proposed that first classifies audio frames as natural sound, noise, or silence using bagged SVM. Rule-based classification is then used to further segment silence frames.
3) The approach involves pre-classifying each windowed portion of the audio clip, extracting normalized feature vectors, and then applying the hybrid classifier. This requires less training data while achieving high accuracy for segmenting into four main audio types.
EFR is a new speech codec that provides enhanced speech quality within the same bandwidth as full rate traffic channels. It improves speech quality and increases radio connection reliability against poor signal quality. Testing showed EFR significantly improved measured speech quality index values compared to full rate, even with poor receiver signal quality. While EFR improved voice quality, some call setup failures occurred due to congestion, indicating the need for higher EFR capacity as most customers now use EFR-compatible handsets.
A Noise Reduction Method Based on Modified Least Mean Square Algorithm of Rea...IRJET Journal
This document presents a modified least mean square (LMS) algorithm to reduce noise in real-time speech signals. The proposed approach modifies the standard LMS algorithm by incorporating a Wiener filter. Experiments are conducted on speech samples from the NOIZEUS database with various types of noise at different signal-to-noise ratios. Objective measures like segmental SNR, log likelihood ratio, Itakura-Saito spectral distance, and cepstrum are used to evaluate the performance of the proposed algorithm compared to the standard LMS algorithm. The results show that the modified LMS algorithm with Wiener filter outperforms the standard LMS algorithm in enhancing the quality of noisy speech signals based on the objective measure values.
IJCER (www.ijceronline.com) International Journal of computational Engineerin...ijceronline
This document summarizes research on using linear predictive coding (LPC) and related techniques for speech recognition and compression. Key points discussed include:
1) LPC is used to compress and encode speech signals for transmission by determining a filter to predict samples from past values, minimizing error. Filter coefficients are encoded and decoded.
2) LPC and PARCOR parameters can characterize phonemes and have potential for speech recognition by analyzing short frames of speech. Recognition rates of 65% for vowels and 94% for consonants were achieved.
3) An LPC-based speech coding system was implemented and tested for mobile radio communications, achieving a bit error rate performance suitable for speech transmission.
IRJET- Emotion recognition using Speech Signal: A ReviewIRJET Journal
This document provides a review of speech emotion recognition techniques. It discusses how speech emotion recognition systems work, including common features extracted from speech like MFCCs and LPC coefficients. Classification techniques used in these systems are also examined, such as DTW, ANN, GMM, and K-NN. The document concludes that speech emotion recognition could be useful for applications requiring natural human-computer interaction, like car systems that monitor driver emotion or educational tutorials that adapt based on student emotion.
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.
55 w 60126-0-tech-brief_keys-to-coherent-acq-successCao Xuân Trình
As demand for data increases, network operators are searching for ways to increase throughput in optical networks beyond 100Gb/s. This requires complex modulation formats that present new challenges for test equipment designers. This document examines key aspects of coherent optical acquisition systems that impact their effectiveness for testing at high data rates, including optimal error vector magnitude (EVM) floor, future-proofing for next generation technologies, and analysis techniques. The digitizer's effective number of bits (ENOB) and use of asynchronous time interleaving are highlighted as important factors for achieving the lowest possible EVM.
55 w 60126-0-tech-brief_keys-to-coherent-acq-successCao Xuân Trình
This document discusses key aspects to consider when choosing a coherent optical acquisition system for testing high-speed fiber optic networks. It covers optimal error vector magnitude (EVM) performance, future-proofing the system for next generation technologies, and analysis techniques for thorough evaluation. Specifically, it emphasizes the importance of oscilloscope bandwidth, sample rate, and effective number of bits for achieving the lowest possible EVM. It also highlights the benefits of a modular oscilloscope system that can be reconfigured to test both current and emerging 100G, 400G, and 1Tb/s networks.
Emotion Recognition based on audio signal using GFCC Extraction and BPNN Clas...ijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Course Content of 6ET3 Digital communication,of B.E. VIth Semester (Electronics & Telecommunication Engineering) of Sant Gadge Baba Amravati University, Maharashtra, India
Speech coding is used to efficiently transmit speech through digital channels by retaining only the information useful to listeners. The LPC-10 standard uses linear predictive coding with 10 coefficients to analyze and synthesize speech. During analysis, it extracts parameters like voicing and pitch from the speech signal. The synthesis process uses these parameters to generate noise or periodic excitation, apply an LPC filter, and control gain. LPC-10 transmits speech at 2.4kbps by coding the 10 LPC coefficients, pitch, voicing, and energy into 54 bits per frame. It enables understandable but unnatural sounding speech and is used for secure voice transmissions.
This document discusses various techniques for speech coding used in digital communication systems. It covers fundamental concepts like sampling theory, quantization, predictive coding, and linear predictive coding (LPC). It then describes specific speech codecs including PCM, ADPCM, CELP, LD-CELP, ACELP, and LPC vocoders. It discusses characteristics of speech coding like being lossy and metrics like SNR and MOS. Finally, it provides details on widely used standards like G.711, G.729, G.723.1, and GSM.
This document provides an overview of random processes, which are used to model information sources in source coding and communications systems. It first defines basic probability concepts like events, sample spaces, and the axioms of probability. It then introduces random variables as mappings from a sample space to real numbers, and random processes as functions of a random variable and time. Common random process models are described, including memoryless, Markov, and stationary processes. Understanding random processes allows analyzing source coding performance based on probabilistic source models.
This document discusses parameters for measuring speech quality in cellular networks. It describes how speech quality is an important aspect of mobile communication quality. Key parameters for measuring speech quality include the Speech Quality Index (SQI), which estimates speech quality as perceived by a human listener. Other parameters discussed include RxQual, Bit Error Rate, Frame Error Rate, and L3 messages. The document also discusses intrusive and non-intrusive methods for measuring speech quality, as well as scales like Mean Opinion Score and Perceptual Evaluation of Speech Quality. Network performance is evaluated using these speech quality measurements and other key performance indicators.
Multirate signal processing and decimation interpolationransherraj
This document is a report on multirate signal processing, decimation, and interpolation. It begins with an introduction that defines multirate signal processing as using signals with different sampling frequencies. It then discusses decimation, which decreases the sampling rate by removing samples, and interpolation, which increases the sampling rate by estimating values between known samples. Applications of multirate signal processing are also discussed, such as digital audio and speech processing. The report concludes that changing the sampling frequency through decimation and interpolation can increase processing efficiency.
The document discusses multi-dimensional approaches to measuring voice quality over packet networks. It describes measuring three key dimensions: delay, echo, and clarity. Delay is one of the most challenging aspects, as real-time voice requires low delay. The document outlines various sources of delay in voice over packet (VoP) networks and techniques for measuring round-trip and one-way delay using specialized instruments and digital signal processing algorithms.
The document discusses GSM network optimization techniques including adjusting parameters for cell selection, power control, and handover control to improve coverage, interference, and handover behavior. It describes the optimization process including initial, primary, and maintenance phases. Key parameters and techniques discussed include enabling features like discontinuous transmission, frequency hopping, and power control as well as adjusting neighbor cell lists, antenna configuration, and frequencies. Drive testing is used to identify problems, verify solutions, and ensure quality.
The document compares the performance of single stage and double stage interleavers in communication systems using turbo codes. A single stage interleaver uses one random interleaver between two convolutional encoders, while a double stage interleaver uses two interleavers in series. The document suggests that a double stage interleaver can improve the bit error rate (BER) of the system compared to a single stage interleaver by further scrambling the input bits. It also provides details on the components of a turbo code system such as convolutional encoders, interleavers, puncturing, and iterative decoding.
This document describes Fracton Technologies and its network optimization solutions. Fracton is an Indian company that focuses on innovations in radio access network optimization for GSM, UMTS, and LTE technologies. It offers a range of network optimization services and has an outstanding track record of optimizing networks for leading mobile operators. Its flagship product is MaxCell, an automated optimization tool that optimizes network performance and quality of experience by tuning cell database parameters based on network configuration and performance data.
IRJET- Segmentation in Digital Signal ProcessingIRJET Journal
1) The document discusses audio signal segmentation techniques in digital signal processing. It analyzes algorithms for segmenting audio signals into homogeneous segments for applications like audio event recognition and speaker identification.
2) A hybrid classification approach is proposed that first classifies audio frames as natural sound, noise, or silence using bagged SVM. Rule-based classification is then used to further segment silence frames.
3) The approach involves pre-classifying each windowed portion of the audio clip, extracting normalized feature vectors, and then applying the hybrid classifier. This requires less training data while achieving high accuracy for segmenting into four main audio types.
EFR is a new speech codec that provides enhanced speech quality within the same bandwidth as full rate traffic channels. It improves speech quality and increases radio connection reliability against poor signal quality. Testing showed EFR significantly improved measured speech quality index values compared to full rate, even with poor receiver signal quality. While EFR improved voice quality, some call setup failures occurred due to congestion, indicating the need for higher EFR capacity as most customers now use EFR-compatible handsets.
A Noise Reduction Method Based on Modified Least Mean Square Algorithm of Rea...IRJET Journal
This document presents a modified least mean square (LMS) algorithm to reduce noise in real-time speech signals. The proposed approach modifies the standard LMS algorithm by incorporating a Wiener filter. Experiments are conducted on speech samples from the NOIZEUS database with various types of noise at different signal-to-noise ratios. Objective measures like segmental SNR, log likelihood ratio, Itakura-Saito spectral distance, and cepstrum are used to evaluate the performance of the proposed algorithm compared to the standard LMS algorithm. The results show that the modified LMS algorithm with Wiener filter outperforms the standard LMS algorithm in enhancing the quality of noisy speech signals based on the objective measure values.
IJCER (www.ijceronline.com) International Journal of computational Engineerin...ijceronline
This document summarizes research on using linear predictive coding (LPC) and related techniques for speech recognition and compression. Key points discussed include:
1) LPC is used to compress and encode speech signals for transmission by determining a filter to predict samples from past values, minimizing error. Filter coefficients are encoded and decoded.
2) LPC and PARCOR parameters can characterize phonemes and have potential for speech recognition by analyzing short frames of speech. Recognition rates of 65% for vowels and 94% for consonants were achieved.
3) An LPC-based speech coding system was implemented and tested for mobile radio communications, achieving a bit error rate performance suitable for speech transmission.
IRJET- Emotion recognition using Speech Signal: A ReviewIRJET Journal
This document provides a review of speech emotion recognition techniques. It discusses how speech emotion recognition systems work, including common features extracted from speech like MFCCs and LPC coefficients. Classification techniques used in these systems are also examined, such as DTW, ANN, GMM, and K-NN. The document concludes that speech emotion recognition could be useful for applications requiring natural human-computer interaction, like car systems that monitor driver emotion or educational tutorials that adapt based on student emotion.
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.
55 w 60126-0-tech-brief_keys-to-coherent-acq-successCao Xuân Trình
As demand for data increases, network operators are searching for ways to increase throughput in optical networks beyond 100Gb/s. This requires complex modulation formats that present new challenges for test equipment designers. This document examines key aspects of coherent optical acquisition systems that impact their effectiveness for testing at high data rates, including optimal error vector magnitude (EVM) floor, future-proofing for next generation technologies, and analysis techniques. The digitizer's effective number of bits (ENOB) and use of asynchronous time interleaving are highlighted as important factors for achieving the lowest possible EVM.
55 w 60126-0-tech-brief_keys-to-coherent-acq-successCao Xuân Trình
This document discusses key aspects to consider when choosing a coherent optical acquisition system for testing high-speed fiber optic networks. It covers optimal error vector magnitude (EVM) performance, future-proofing the system for next generation technologies, and analysis techniques for thorough evaluation. Specifically, it emphasizes the importance of oscilloscope bandwidth, sample rate, and effective number of bits for achieving the lowest possible EVM. It also highlights the benefits of a modular oscilloscope system that can be reconfigured to test both current and emerging 100G, 400G, and 1Tb/s networks.
Emotion Recognition based on audio signal using GFCC Extraction and BPNN Clas...ijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
A Novel, Robust, Hierarchical, Text-Independent Speaker Recognition TechniqueCSCJournals
Automatic speaker recognition system is used to recognize an unknown speaker among several reference speakers by making use of speaker-specific information from their speech. In this paper, we introduce a novel, hierarchical, text-independent speaker recognition. Our baseline speaker recognition system accuracy, built using statistical modeling techniques, gives an accuracy of 81% on the standard MIT database and our baseline gender recognition system gives an accuracy of 93.795%. We then propose and implement a novel state-space pruning technique by performing gender recognition before speaker recognition so as to improve the accuracy/timeliness of our baseline speaker recognition system. Based on the experiments conducted on the MIT database, we demonstrate that our proposed system improves the accuracy over the baseline system by approximately 2%, while reducing the computational time by more than 30%.
International Journal of Engineering Research and Development (IJERD)IJERD Editor
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Describe The Main Functions Of Each Layer In The Osi Model...Amanda Brady
Tone injection is a technique used to increase the constellation size of a signal constellation. It works by mapping each point in the original constellation to multiple equivalent points in an expanded constellation. This allows for embedding additional information by substituting points, improving spectral efficiency. However, it also increases implementation complexity and may degrade performance due to increased decision regions. Tone injection is useful for applications requiring high data rates within bandwidth constraints.
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
IJERD (www.ijerd.com) International Journal of Engineering Research and Devel...IJERD Editor
This document summarizes and compares several techniques for enhancing the intelligibility of speech signals corrupted by noise. It describes single channel techniques like spectral subtraction, spectral subtraction with oversubtraction, and nonlinear spectral subtraction. It also covers multi-channel techniques such as adaptive noise cancellation and multisensory beamforming. Additionally, it discusses spectral subtraction using adaptive averaging, noise reduction using enhanced Wiener filtering, and other adaptive neuro-fuzzy techniques for speech enhancement. The goal of these techniques is to improve the quality and intelligibility of noisy speech signals.
This document provides information on various measurement solutions for the automotive industry, including:
1) Systems for measuring engine-related parameters like combustion chamber volume, analyzing engine noise and vibrations, and inspecting transmission components.
2) Solutions for testing vehicle bodies, including analyzing vibration modes and locating sound sources.
3) Methods for evaluating interior comfort, such as measuring sound absorption and performing psychoacoustic assessments.
4) A system for testing muffler acoustic performance.
The document also provides information on related products like multi-channel analyzers, sensors, and anechoic chambers.
This document provides information on various measurement solutions for the automotive industry, including:
1) Systems for measuring engine-related parameters like combustion chamber volume, analyzing engine noise and vibrations, and inspecting transmission components.
2) Solutions for testing vehicle bodies, including analyzing vibration modes and locating sound sources.
3) Methods for evaluating interior comfort, such as assessing sound quality and testing absorption of interior materials.
4) A system for measuring the sound absorption of exhaust mufflers.
The document describes the basic principles and components of a speech recognition system. It discusses the two main modules of feature extraction and feature matching. Feature extraction involves extracting representative data from voice signals, while feature matching compares extracted features to stored reference models to identify unknown speakers. The system has two phases - an enrollment phase where reference models are built for each registered speaker, and a testing phase where input speech is matched to models to make a recognition decision. Speech recognition has applications in security systems for verifying user identities.
An effective evaluation study of objective measures using spectral subtractiv...eSAT Journals
Abstract
Unwanted noises have a negative influence over communication because it disturbs the conversation and make the communication impossible. Speech enhancement algorithms are used for improving the quality and intelligibility or to reduce listener fatigues. Assessment of speech quality can be done by using either subjective listening test or objective quality measure. Evaluation of several objective measures with the speech processed by enhancement algorithms has been performed but these having limitations to assess original speech signal. This paper represents the study of speech quality measures and compute the values used for regression analyses of the objective measures evaluation study using spectral subtraction algorithm based enhanced speech signal.
Keywords: MOS, ITU-T (P.835), SNRseg, log- likelihood ratio and itakura-saito.
Similar to Automatic Sound Signals Quality Estimation (20)
This document is a user manual for Sevana AQuA - Audio Quality Analyzer 8.x. It introduces AQuA as a tool for intrusive perceptual voice quality analysis that allows comparing two audio files and testing voice quality loss between a reference and degraded file. The manual describes AQuA's functionality, requirements, testing parameters, scientific background, perceptual modeling, command line usage, and provides examples of comparing audio files and analyzing reasons for voice quality loss.
QualTest mobile test probe for VoIP and mobile call testing and monitoringSevana Oü
Sevana QualTest is a mobile test probe application that checks
current network conditions and estimates voice quality in mobile and VoIP networks
The platform is designed for end-to-end and single-end call testing, as well as for gathering and analyzing call audio quality metrics. Measure network metrics for VoIP calls and use waveform analysis to correlate audio problems with network conditions.
Real-time monitoring of 5G network. Reliable MOS and other KPIs for messenger-to-messenger and voice calls. Automated mobile-to-mobile testing in 5G networks. Flexible integration and drive testing.
This utility calculates MOS scores for audio streams in .pcap files, optionally decoding the audio to .wav files. It runs on Linux, macOS, and OpenWRT, requires no database, and supports several common codecs. The user provides a .pcap file path and can choose json output or audio saving. The utility then extracts and analyzes RTP streams, calculating MOS scores and statistics and printing the results.
Messenger-to-messenger testing. Skype call quality test.Sevana Oü
This document discusses using Sevana tools to test call quality between mobile devices using Skype. Specifically, it proposes a demonstration of a Skype-to-Skype call quality test using regular Android Skype apps on mobile phones. The test would record call audio and analyze it using Sevana's AQUA and PVQA tools to generate MOS scores and other quality metrics. It outlines the typical test setup and flow, including host machine, mobile phone, cable adapter, and test management via a backend server.
Administration manual for Sevana Voice Quality Monitoring solution based on Asterisk PBX. This solutions makes end-to-end voice quality testing and monitoring easy. Various test scenarios for echo or conference birdge testing are already included. AQuA and PVQA impairments analysis together with full VoIP statistics make it suitable for use in any type of network.
Integrate QualTest GSM with desktop or Raspberry Pi. Application receives notification from QualTest test probes about call events, copies recorded calls to desktop, limits time of call, runs pvqa and aqua utilities to estimate voice quality.
Hardware requirements.
QualTest GSM is an Android application that turns your mobile phone into mobile test probe that can make mobile-to-mobile test calls returing AQuA and PVQA call quality metrics including MOS score and audio impairments information.
The document discusses a mobile-to-mobile call quality testing automation solution called Sevana QualTest. It allows for unlimited automated mobile-to-mobile tests without human involvement, saving time compared to manual testing. Key features include active and passive call quality tests using non-rooted Android phones, test result uploads and sharing, and compatibility with all operating systems. Potential use cases include 5G and other mobile network testing, service assurance through quality monitoring, and RAN optimization to improve quality management while reducing call quality testing costs through automation.
Sevana real-time rtp analysis for mobile operatorsSevana Oü
The document describes Sevana's real-time RTP analysis solution for mobile networks. The solution provides call quality monitoring through metrics like MOS and detection of impairments. It allows analysis of call quality for different carriers to evaluate performance, select partners, and identify issues. The solution integrates with existing systems and provides real-time quality monitoring to help optimize networks and prevent churn.
Real time call quality analysis for mobile operatorsSevana Oü
Real waveform analysis for mobile service and solution providers to easily enable:
- Silent call detection
- Audio impairments patterns analysis
- Problem root cause detection
- Fraud detection
- Customized RTP stream interaction
Sevana QualTest is a mobile test probe application which checks current network conditions and estimates voice quality in mobile networks. QualTest is a mobile application for Android
powered devices that can work in both VoIP and cellular networks.
Powerful tool to non-intrusively evaluate
voice quality. Quick and easy setup up of single-ended
voice quality testing. Objective MOS score prediction. Call quality impairments detection.
Sevana PVQA Server is a multipurpose tool for call quality analysis. One can use it as core of their quality analysis and monitoring system to detect and investigate QoE and QoS related issues. PVQA server combines the force of audio waveform analysis using PVQA technology with continuous network monitoring.
Sevana AQuA - Your powerful tool for perceptual voice and audio quality analysis:
• Quick and easy setup up of end-to-end voice quality testing
• Reliable objective MOS score
• Comprehensive waveform analysis
The document describes a real-time RTP call quality monitoring solution using waveform analysis that provides several benefits:
It allows service providers to efficiently manage voice/audio quality in a non-intrusive way based on analyzing the actual media content. This can save on operating expenses by reducing unnecessary payments to partners and prevent revenue loss from customer churn due to poor call quality. The solution provides reliable objective quality metrics and insight into the root causes of quality issues to help justify infrastructure investments. It has high performance and scalability to handle thousands of simultaneous calls through an asynchronous architecture utilizing multiple CPU cores.
This document contains the user manual for Sevana AQuA - Audio Quality Analyzer 7.x. It describes the functionality, requirements, and parameters for using AQuA to compare wav files and test voice quality. Key features include comparing reference files to degraded files, reporting quality scores, analyzing reasons for quality loss, and visualizing signal spectrums. The document also provides details on AQuA's command line parameters and examples of usage.
This document discusses objective quality measurement tools for evaluating voice quality over mobile networks.
It introduces AQuA, an artificial intelligence-based tool that provides more accurate quality scores than PESQ and POLQA by considering factors like codecs, technologies, impairments and languages.
AQuA and PVQA can analyze drive test recordings to generate MOS scores and identify issues like packet loss, hardware problems or network impairments that caused quality degradation.
Tables compare features of PESQ, POLQA and AQuA, showing AQuA can evaluate all audio types and technologies more comprehensively than the other tools.
Getting the Most Out of ScyllaDB Monitoring: ShareChat's TipsScyllaDB
ScyllaDB monitoring provides a lot of useful information. But sometimes it’s not easy to find the root of the problem if something is wrong or even estimate the remaining capacity by the load on the cluster. This talk shares our team's practical tips on: 1) How to find the root of the problem by metrics if ScyllaDB is slow 2) How to interpret the load and plan capacity for the future 3) Compaction strategies and how to choose the right one 4) Important metrics which aren’t available in the default monitoring setup.
[OReilly Superstream] Occupy the Space: A grassroots guide to engineering (an...Jason Yip
The typical problem in product engineering is not bad strategy, so much as “no strategy”. This leads to confusion, lack of motivation, and incoherent action. The next time you look for a strategy and find an empty space, instead of waiting for it to be filled, I will show you how to fill it in yourself. If you’re wrong, it forces a correction. If you’re right, it helps create focus. I’ll share how I’ve approached this in the past, both what works and lessons for what didn’t work so well.
"Frontline Battles with DDoS: Best practices and Lessons Learned", Igor IvaniukFwdays
At this talk we will discuss DDoS protection tools and best practices, discuss network architectures and what AWS has to offer. Also, we will look into one of the largest DDoS attacks on Ukrainian infrastructure that happened in February 2022. We'll see, what techniques helped to keep the web resources available for Ukrainians and how AWS improved DDoS protection for all customers based on Ukraine experience
As AI technology is pushing into IT I was wondering myself, as an “infrastructure container kubernetes guy”, how get this fancy AI technology get managed from an infrastructure operational view? Is it possible to apply our lovely cloud native principals as well? What benefit’s both technologies could bring to each other?
Let me take this questions and provide you a short journey through existing deployment models and use cases for AI software. On practical examples, we discuss what cloud/on-premise strategy we may need for applying it to our own infrastructure to get it to work from an enterprise perspective. I want to give an overview about infrastructure requirements and technologies, what could be beneficial or limiting your AI use cases in an enterprise environment. An interactive Demo will give you some insides, what approaches I got already working for real.
Keywords: AI, Containeres, Kubernetes, Cloud Native
Event Link: https://meine.doag.org/events/cloudland/2024/agenda/#agendaId.4211
Must Know Postgres Extension for DBA and Developer during MigrationMydbops
Mydbops Opensource Database Meetup 16
Topic: Must-Know PostgreSQL Extensions for Developers and DBAs During Migration
Speaker: Deepak Mahto, Founder of DataCloudGaze Consulting
Date & Time: 8th June | 10 AM - 1 PM IST
Venue: Bangalore International Centre, Bangalore
Abstract: Discover how PostgreSQL extensions can be your secret weapon! This talk explores how key extensions enhance database capabilities and streamline the migration process for users moving from other relational databases like Oracle.
Key Takeaways:
* Learn about crucial extensions like oracle_fdw, pgtt, and pg_audit that ease migration complexities.
* Gain valuable strategies for implementing these extensions in PostgreSQL to achieve license freedom.
* Discover how these key extensions can empower both developers and DBAs during the migration process.
* Don't miss this chance to gain practical knowledge from an industry expert and stay updated on the latest open-source database trends.
Mydbops Managed Services specializes in taking the pain out of database management while optimizing performance. Since 2015, we have been providing top-notch support and assistance for the top three open-source databases: MySQL, MongoDB, and PostgreSQL.
Our team offers a wide range of services, including assistance, support, consulting, 24/7 operations, and expertise in all relevant technologies. We help organizations improve their database's performance, scalability, efficiency, and availability.
Contact us: info@mydbops.com
Visit: https://www.mydbops.com/
Follow us on LinkedIn: https://in.linkedin.com/company/mydbops
For more details and updates, please follow up the below links.
Meetup Page : https://www.meetup.com/mydbops-databa...
Twitter: https://twitter.com/mydbopsofficial
Blogs: https://www.mydbops.com/blog/
Facebook(Meta): https://www.facebook.com/mydbops/
What is an RPA CoE? Session 1 – CoE VisionDianaGray10
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Topics covered:
• The role of a steering committee
• How do the organization’s priorities determine CoE Structure?
Speaker:
Chris Bolin, Senior Intelligent Automation Architect Anika Systems
QA or the Highway - Component Testing: Bridging the gap between frontend appl...zjhamm304
These are the slides for the presentation, "Component Testing: Bridging the gap between frontend applications" that was presented at QA or the Highway 2024 in Columbus, OH by Zachary Hamm.
From Natural Language to Structured Solr Queries using LLMsSease
This talk draws on experimentation to enable AI applications with Solr. One important use case is to use AI for better accessibility and discoverability of the data: while User eXperience techniques, lexical search improvements, and data harmonization can take organizations to a good level of accessibility, a structural (or “cognitive” gap) remains between the data user needs and the data producer constraints.
That is where AI – and most importantly, Natural Language Processing and Large Language Model techniques – could make a difference. This natural language, conversational engine could facilitate access and usage of the data leveraging the semantics of any data source.
The objective of the presentation is to propose a technical approach and a way forward to achieve this goal.
The key concept is to enable users to express their search queries in natural language, which the LLM then enriches, interprets, and translates into structured queries based on the Solr index’s metadata.
This approach leverages the LLM’s ability to understand the nuances of natural language and the structure of documents within Apache Solr.
The LLM acts as an intermediary agent, offering a transparent experience to users automatically and potentially uncovering relevant documents that conventional search methods might overlook. The presentation will include the results of this experimental work, lessons learned, best practices, and the scope of future work that should improve the approach and make it production-ready.
What is an RPA CoE? Session 2 – CoE RolesDianaGray10
In this session, we will review the players involved in the CoE and how each role impacts opportunities.
Topics covered:
• What roles are essential?
• What place in the automation journey does each role play?
Speaker:
Chris Bolin, Senior Intelligent Automation Architect Anika Systems
Connector Corner: Seamlessly power UiPath Apps, GenAI with prebuilt connectorsDianaGray10
Join us to learn how UiPath Apps can directly and easily interact with prebuilt connectors via Integration Service--including Salesforce, ServiceNow, Open GenAI, and more.
The best part is you can achieve this without building a custom workflow! Say goodbye to the hassle of using separate automations to call APIs. By seamlessly integrating within App Studio, you can now easily streamline your workflow, while gaining direct access to our Connector Catalog of popular applications.
We’ll discuss and demo the benefits of UiPath Apps and connectors including:
Creating a compelling user experience for any software, without the limitations of APIs.
Accelerating the app creation process, saving time and effort
Enjoying high-performance CRUD (create, read, update, delete) operations, for
seamless data management.
Speakers:
Russell Alfeche, Technology Leader, RPA at qBotic and UiPath MVP
Charlie Greenberg, host
"What does it really mean for your system to be available, or how to define w...Fwdays
We will talk about system monitoring from a few different angles. We will start by covering the basics, then discuss SLOs, how to define them, and why understanding the business well is crucial for success in this exercise.
"$10 thousand per minute of downtime: architecture, queues, streaming and fin...Fwdays
Direct losses from downtime in 1 minute = $5-$10 thousand dollars. Reputation is priceless.
As part of the talk, we will consider the architectural strategies necessary for the development of highly loaded fintech solutions. We will focus on using queues and streaming to efficiently work and manage large amounts of data in real-time and to minimize latency.
We will focus special attention on the architectural patterns used in the design of the fintech system, microservices and event-driven architecture, which ensure scalability, fault tolerance, and consistency of the entire system.
In the realm of cybersecurity, offensive security practices act as a critical shield. By simulating real-world attacks in a controlled environment, these techniques expose vulnerabilities before malicious actors can exploit them. This proactive approach allows manufacturers to identify and fix weaknesses, significantly enhancing system security.
This presentation delves into the development of a system designed to mimic Galileo's Open Service signal using software-defined radio (SDR) technology. We'll begin with a foundational overview of both Global Navigation Satellite Systems (GNSS) and the intricacies of digital signal processing.
The presentation culminates in a live demonstration. We'll showcase the manipulation of Galileo's Open Service pilot signal, simulating an attack on various software and hardware systems. This practical demonstration serves to highlight the potential consequences of unaddressed vulnerabilities, emphasizing the importance of offensive security practices in safeguarding critical infrastructure.
Lee Barnes - Path to Becoming an Effective Test Automation Engineer.pdfleebarnesutopia
So… you want to become a Test Automation Engineer (or hire and develop one)? While there’s quite a bit of information available about important technical and tool skills to master, there’s not enough discussion around the path to becoming an effective Test Automation Engineer that knows how to add VALUE. In my experience this had led to a proliferation of engineers who are proficient with tools and building frameworks but have skill and knowledge gaps, especially in software testing, that reduce the value they deliver with test automation.
In this talk, Lee will share his lessons learned from over 30 years of working with, and mentoring, hundreds of Test Automation Engineers. Whether you’re looking to get started in test automation or just want to improve your trade, this talk will give you a solid foundation and roadmap for ensuring your test automation efforts continuously add value. This talk is equally valuable for both aspiring Test Automation Engineers and those managing them! All attendees will take away a set of key foundational knowledge and a high-level learning path for leveling up test automation skills and ensuring they add value to their organizations.
Lee Barnes - Path to Becoming an Effective Test Automation Engineer.pdf
Automatic Sound Signals Quality Estimation
1. Automatic Sound Signals Quality
Estimation
Sevana Oy
Endre Domiczi
Mobile: +372 53485178
E-mail: ceo@sevana.fi
2. Automatic Sound Signals Quality Estimation
• General outline and the basic components of the system – signal,
synchronization and analytical components
• Test signal (including statistic speech model) and sound perception models
• Adequacy of analytical estimations based on the results of the comparison
of received analytical and subjective MOS estimations
• Acoustic model further developments
• Available software
• System applications
3. Review of existing quality estimation methods
• AI (Articulation Index). The idea is that the whole frequency range of
speech signal is divided into 20 bands and the signal/noise ratio is
determined within the band. The band broad is defined in such a way, that
every band contributes equally in speech perception. The signal/noise ratio
is calculated within every band. Articulation index is supposed to be equal
the weighted total of the band values. The articulation index does not take
into account the properties of hearing and speech production, although it
directs toward speech signal.
• SII (Speech Intelligibility Index) is the evolution of AI method. The
American Standard ANSI S3.5-1997 includes the speech intelligibility index.
It provides for 4 measuring procedures on different band groups: 21 critical
bands, 18 one third-octave bands, 17 equal by their contribution critical
bands and 6 octave bands. The signal/noise ratio is calculated within every
band and the total SII coefficient, ranged from 0 to 1 is computed. The
speech intelligibility index takes into account only the properties of hearing,
not speech production.
4. Review of existing quality estimation methods
• STI (Speech Transmission Index). We may approximately consider speech signal
as broadband signal modulated by low-frequency signal. Articulation speed
determines modulation frequency. When modulation depth decreases, speech signal
becomes similar to noise and its intelligibility decreases. Accordingly, intelligibility
decrease can be estimated according to modulation depth decrease also. Whole
speech range is divided into 7octave bands. An octave noise signal is the input. The
test signal intensity distribution agrees with the distribution of speech signal
intensities. The modulating signal frequencies vary from 0.5 to 12.5 Hz with one-third-
octave interval (14 frequencies in all). The STI measuring method is stated in the
International standard IEC 268-16.
• RATSI/STIPA (Rapid Speech Transmission Index). The STI method needs a lot of
measuring procedures and calculations. A simplified method was developed, which
provides for measuring only in 2 bands with 5 modulation frequencies and reduces the
number of measuring procedures and calculations. For good intelligibility RASTI
values must be not less than 0.6.
Speech transmission index as well as rapid speech transmission index imitates
speech production process by means of noise model, but to take into account the
properties of speech production and hearing in such a way is far from optimum.
5. Review of existing quality estimation methods
• C50 (factor of clearness) determines sound clearness and clarity. It is
computed as near echo/far echo ratio. The method is based on the fact, that
echo reduces signal intelligibility. The near echo/far echo ratios in several
frequency bands are calculated. They consider near echo (less than 33 ms)
as useful signal and far echo (more than 33 ms) as disturbing signal. The
factor of clearness takes into account only the one kind of the possible
distortions and it is worth to apply it only as one of the speech quality
estimations.
The need to develop new methods and to improve existing ones is caused by
desire to bring together objective and subjective estimation of quality and to
explicitly use in such systems our knowledge about hearing and speech
production.
To use arbitrary or particularized signal as a source signal depends on the
estimation purpose (speech intelligibility evaluation, sound reproduction quality,
quality estimation of speech, transmitted through intercommunication channels,
etc.) and allows increasing estimation objectivity.
6. Introduced System General Concept
Block of signals
Device
Synchronizer
under test
result
Analytical
Bank of
module
signals
Test signal
generator Estimation block
Generator of test signals allows sound signal forming according to one of the sound flow
models. It can be either a particularized set of sound signals or a signal, received in output
of statistical speech model. Generator’s signal can either be saved for follow-up usage or
be exposed to processing and estimation. Bank of signals stores sound data, received as
a result of signals’ generator work or from some external sources.
Accordingly, an input of estimation block is a signal of generator directly or one of the bank
of signals. Test signal is the input of the synchronizer or of the device under test, which
can be for example, a vocoder or a communication channel. The output signal of the
device under test is an input of synchronizer also.
The synchronizer matches in time an initial signal and a processed signal. The
synchronized signals in chunks input in analytical module, which determines the degree of
similarity for signals and issues the quality estimation as the measure of similarity between
the initial and the processed signals.
7. Implementation
Algorithms described are implemented in Sevana Audio Codecs Analyzer for
vocoder quality estimation and comparison of external initial signals and signals
under test.
As the external signals arbitrary signals recorded with the sampling frequency of
8 kHz and the capacity of samples equal 16 bits can be used. Supposed, the
signal under test is received from an initial signal as a result of some
transformations (for example, compression/restoration, transmission through
communication channels, filtration).
As internal initial signals (i.e. signals, which the user of the program has no
access to) the signals generated according to the proprietary noise model and
the signals, generated on the basis of the statistic model.
Internal input signals of sound data to the system, are implemented as DLL. One
can use both DLLs provided within Sevana Audio Codecs Analyzer or developed
by the others. The signal processed by means of methods contained in DLL is
consider as the signal under test and is exposed to the proprietary quality
estimation procedure.
8. Advantages
• It is a universal tool since it allows judging the quality of signals from
various sources and processed in different ways;
• One can optimize quality estimation signal depending on the
purposes:
– in speed (for example, it is possible to receive rough estimation
quickly);
– in signal type (using different bands for speech signals and
sound signals in general);
• Resulting estimations correlate well with that of МОS;
• Quality estimations received for speech signals can be translated in
values of various kinds of intelligibility.
9. Test Results
Noise model
Statistic model PhRT
Codec MOS Minimal Reduced Complete
- Vc - Vc - Vc - Vc - Vc
A-Law 4,10 4,79 4,73 4,78 4,78 4,78 4,78 4,79 4,80 4,80 4,84
Mu-Law 4,10 4,79 4,84 4,77 4,77 4,77 4,78 4,78 4,79 4,79 4,82
G.723.6.3 3,90 4,25 4,48 4,21 4,29 4,22 4,33 4,15 4,04 4,08 3,95
GSM.6.10 3,70 3,20 1,99 3,01 1,65 3,04 1,78 4,22 3,66 4,01 3,21
G.723.5.3 3,65 4,23 4,44 4,18 4,27 4,19 4,32 4,14 4,04 4,06 3,93
The table above represents quality estimations of several standard vocoders, received on various test signals
using the proprietary method and Sevana Audio Codecs Analyzer. The table contains MOS estimations for
comparison.
Estimations under the assumption, that bands are of equal probability, are in the column with «-» symbol and
the estimation received under taking into account the coefficients of importance are in the column with «Vc»
10. Applications: quality estimation of sound transmission
through telephone network of general use
Telephone network
of general use
Reception
Transmission
Telephone Telephone
or modem or modem
Signal under test
Initial signal
Server of sound
quality estimation
The picture represents applying the method described above for quality estimation of sound transmission through telephone network
of general use. The given scheme is applicable both for local and for long-distance connections.
Server of sound quality estimation generates an initial signal (or chooses among signals prepared before) and transfers it to one of
the telephone subscribers taking part in the testing. The subscriber received the signal establishes a standard connection with the
second subscriber and reproduced the initial signal. The second subscriber records the receiving sound signal and transfers it to the
sound quality estimation server.
The sound quality estimation server compares the initial and the test signals according to the suggested method and gives the
quality estimation of the signal transferred through the telephone network. The received estimation can be used for improving the
subscriber service, deciding about the necessity of equipment changing or setting (both on the side of the subscriber and on the
station), as advertising and others.
11. Applications: quality estimation of sound transferred
through IP-network
IP-network
ion So
ss un
mi d
ns sig
tra na
al l re
ign ce
pti
ls on
itia
In
VoIP-Server 1 VoIP-Server 2
Initial signal Signal under test
Sound quality
estimation server
Similarly to telephone network quality estimation of sound transferred through IP-network
is performed. It differs from the previous application in the way of transferring the initial and
the signal under test from the sound quality estimation server to subscribers and in the
way of data transfer between subscribers.
Quality estimations received can be used for choosing codecs used in VoIP-connection
and when choosing operators, providing IP-telephony services.
12. Applications: sound quality estimation of cellular and
satellite connection
Satellite com m unication
Initial signal
transm ission Initial signal
reception
C ellular network
Sm artphone or
Sm artphone or
m obile phone
m obile phone
Initial signal Signal under test
Sound quality estim ation srever
The introduced method and software can be effectively used for quality
estimation of cellular and satellite connections. Received estimations
subscribers use for choosing operators and telephone models and operators use
for optimization of base station locations.
13. Applications: quality estimation of systems and
algorithms (methods) for sound data compression
Workstations of developers and testers
of systems of sound data compression
Initial and test signals,
quality estimations
Sound data base
Signals
Initial signals
under test
Quality estimations
Sound quality estimation server
Every codec version (or codec with a set of parameters) requires estimation and comparison with the
analogs. Every developer can refer to sound sample base, compress and restore a signal and receive
objective quality estimation of the codec work.
Such a system allows managing the codec developing process and the optimization of their parameters
much more effectively. Ultimate consumer will be able to receive not just functioning, but optimal
algorithm.
14. Applications: rooms’ sound quality estimation
Distribution system
Microphone
Speaker
al
ign
s
ial
Init
Sound system 1 Sound system
Signals under test
Sound quality estimation server
In this case initial signal is a signal from the microphone located opposite the speaker and
signals under test are those from microphones located in different parts of the room, in
places where hearers and sound reproducing equipment are located.
The received estimations can be used for optimization of the location of sound reproducing
equipment, furniture and spectators’ places.
15. Further Development
• Integration with existing Quality of Service and Quality of Experience systems to increase their
functionality and enrich test impact.
• Test signal model improvement. Here the noise model can be supplied with a set of multiband
modulated noise signals; enrich the set of data and algorithms of the statistic speech model,
increase the number of prepared test signals (such as records of PhRT);
• Development of more upgraded algorithms of synchronization, based, for example, on
coincidence of maximums in signal energy spectrums;
• Acoustic model modernization with taking into account masking effects and the fact that pure
tones and band noise cause difference in hearing;
•
• Signal comparison scheme modernization. Current distance measure can be more accurate for
strongly different signals. For higher universality of the system it is desired to use the correlation
analysis methods for comparison;
• Solve a number of practical problems the systems requires the possibility to work with
multichannel (Stereo-, Quadro-, etc.) and to receive immediate quality estimations;
• Absolutely correct translation of the objective estimations into MOS estimation values requires
further experimental researches.