This document presents a frequency dependent spectral subtraction method for speech enhancement. The standard spectral subtraction technique advocates subtracting the noise spectrum estimate uniformly over the entire speech spectrum. However, real-world noise is mostly colored and affects speech differently at various frequencies. The proposed method takes this into account by performing noise subtraction in a frequency dependent manner. It divides the spectrum into multiple bands and subtracts noise non-uniformly across the different bands based on the noise characteristics. This leads to improved speech quality and reduced musical noise compared to conventional spectral subtraction. The method is evaluated objectively using measures like perceptual evaluation of speech quality and computational complexity is also considered.
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
Estimating the quality of digitally transmitted speech over satellite communi...Alexander Decker
This document discusses methods for estimating the quality of digitally transmitted speech over satellite
communication channels. It proposes a methodology that divides the speech signal into segments and calculates the
segmental signal-to-noise ratio (SNR) for each segment. This allows assessing changes in speech quality over time
through statistical modeling. It also considers the spectral characteristics of the speech signal, providing a better
measure than just signal power. The document reviews other existing objective and subjective methods for quality
estimation and argues that the proposed segmental SNR approach has advantages over these other approaches.
Performance estimation based recurrent-convolutional encoder decoder for spee...karthik annam
This document discusses a proposed Recurrent-Convolutional Encoder-Decoder (R-CED) network for speech enhancement. The R-CED network aims to overcome challenges with existing methods by estimating the a priori and posteriori signal-to-noise ratios to separate noise from speech. The R-CED consists of convolutional layers with increasing and decreasing numbers of filters to encode and decode features. Performance will be evaluated using metrics like PESQ, STOI, CER, MSE, SNR, and SDR. The proposed method aims to improve speech enhancement accuracy and recover enhanced speech quality compared to other techniques.
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.
Audio Noise Removal – The State of the Artijceronline
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.
In the present-day communications speech signals get contaminated due to
various sorts of noises that degrade the speech quality and adversely impacts
speech recognition performance. To overcome these issues, a novel approach
for speech enhancement using Modified Wiener filtering is developed and
power spectrum computation is applied for degraded signal to obtain the
noise characteristics from a noisy spectrum. In next phase, MMSE technique
is applied where Gaussian distribution of each signal i.e. original and noisy
signal is analyzed. The Gaussian distribution provides spectrum estimation
and spectral coefficient parameters which can be used for probabilistic model
formulation. Moreover, a-priori-SNR computation is also incorporated for
coefficient updation and noise presence estimation which operates similar to
the conventional VAD. However, conventional VAD scheme is based on the
hard threshold which is not capable to derive satisfactory performance and a
soft-decision based threshold is developed for improving the performance of
speech enhancement. An extensive simulation study is carried out using
MATLAB simulation tool on NOIZEUS speech database and a comparative
study is presented where proposed approach is proved better in comparison
with existing technique.
SPEECH ENHANCEMENT USING KERNEL AND NORMALIZED KERNEL AFFINE PROJECTION ALGOR...sipij
This document discusses speech enhancement using kernel and normalized kernel affine projection algorithms. It begins with an abstract that outlines using kernel adaptive filters in reproducing kernel Hilbert space to improve speech quality and intelligibility over conventional adaptive filters. It then provides background on adaptive filtering and its applications in noise cancellation. The document discusses challenges in speech enhancement like non-stationary noise and tradeoffs between noise removal and speech distortion. It describes using adaptive filters and kernel adaptive filters to enhance speech for applications like hearing aids and speech recognition.
This document discusses using deep neural networks for speech enhancement by finding a mapping between noisy and clean speech signals. It aims to handle a wide range of noises by using a large training dataset with many noise/speech combinations. Techniques like global variance equalization and dropout are used to improve generalization. Experimental results show improvements over MMSE techniques, with the ability to suppress nonstationary noise and avoid musical artifacts. The introduction provides background on speech enhancement, recognition using HMMs and other models, and the role of deep learning advances.
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
Estimating the quality of digitally transmitted speech over satellite communi...Alexander Decker
This document discusses methods for estimating the quality of digitally transmitted speech over satellite
communication channels. It proposes a methodology that divides the speech signal into segments and calculates the
segmental signal-to-noise ratio (SNR) for each segment. This allows assessing changes in speech quality over time
through statistical modeling. It also considers the spectral characteristics of the speech signal, providing a better
measure than just signal power. The document reviews other existing objective and subjective methods for quality
estimation and argues that the proposed segmental SNR approach has advantages over these other approaches.
Performance estimation based recurrent-convolutional encoder decoder for spee...karthik annam
This document discusses a proposed Recurrent-Convolutional Encoder-Decoder (R-CED) network for speech enhancement. The R-CED network aims to overcome challenges with existing methods by estimating the a priori and posteriori signal-to-noise ratios to separate noise from speech. The R-CED consists of convolutional layers with increasing and decreasing numbers of filters to encode and decode features. Performance will be evaluated using metrics like PESQ, STOI, CER, MSE, SNR, and SDR. The proposed method aims to improve speech enhancement accuracy and recover enhanced speech quality compared to other techniques.
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.
Audio Noise Removal – The State of the Artijceronline
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.
In the present-day communications speech signals get contaminated due to
various sorts of noises that degrade the speech quality and adversely impacts
speech recognition performance. To overcome these issues, a novel approach
for speech enhancement using Modified Wiener filtering is developed and
power spectrum computation is applied for degraded signal to obtain the
noise characteristics from a noisy spectrum. In next phase, MMSE technique
is applied where Gaussian distribution of each signal i.e. original and noisy
signal is analyzed. The Gaussian distribution provides spectrum estimation
and spectral coefficient parameters which can be used for probabilistic model
formulation. Moreover, a-priori-SNR computation is also incorporated for
coefficient updation and noise presence estimation which operates similar to
the conventional VAD. However, conventional VAD scheme is based on the
hard threshold which is not capable to derive satisfactory performance and a
soft-decision based threshold is developed for improving the performance of
speech enhancement. An extensive simulation study is carried out using
MATLAB simulation tool on NOIZEUS speech database and a comparative
study is presented where proposed approach is proved better in comparison
with existing technique.
SPEECH ENHANCEMENT USING KERNEL AND NORMALIZED KERNEL AFFINE PROJECTION ALGOR...sipij
This document discusses speech enhancement using kernel and normalized kernel affine projection algorithms. It begins with an abstract that outlines using kernel adaptive filters in reproducing kernel Hilbert space to improve speech quality and intelligibility over conventional adaptive filters. It then provides background on adaptive filtering and its applications in noise cancellation. The document discusses challenges in speech enhancement like non-stationary noise and tradeoffs between noise removal and speech distortion. It describes using adaptive filters and kernel adaptive filters to enhance speech for applications like hearing aids and speech recognition.
This document discusses using deep neural networks for speech enhancement by finding a mapping between noisy and clean speech signals. It aims to handle a wide range of noises by using a large training dataset with many noise/speech combinations. Techniques like global variance equalization and dropout are used to improve generalization. Experimental results show improvements over MMSE techniques, with the ability to suppress nonstationary noise and avoid musical artifacts. The introduction provides background on speech enhancement, recognition using HMMs and other models, and the role of deep learning advances.
This document provides an overview of speech recognition systems and recent progress in the field. It discusses different types of speech recognition including isolated word, connected word, continuous speech, and spontaneous speech. Various techniques used in speech recognition are also summarized, such as simulated evolutionary computation, artificial neural networks, fuzzy logic, Kalman filters, and Hidden Markov Models. The document reviews several papers published between 2004-2012 that studied speech recognition methods including using dynamic spectral subband centroids, Kalman filters, biomimetic computing techniques, noise estimation, and modulation filtering. It concludes that Hidden Markov Models combined with MFCC features provide good recognition results and are suitable for large vocabulary, speaker-independent, continuous speech recognition.
A GAUSSIAN MIXTURE MODEL BASED SPEECH RECOGNITION SYSTEM USING MATLABsipij
1) The document describes the development of a speaker-dependent speech recognition system using MATLAB. It uses Gaussian mixture models for acoustic modeling and mel-frequency cepstral coefficients for feature extraction.
2) The system is designed to recognize isolated digits 0-9. Voice activity detection is performed to detect segments of speech. Various windowing functions are evaluated to reduce spectral leakage during feature extraction.
3) 13 MFCCs plus energy are extracted from each 30ms frame with 10ms shift. The first and second derivatives are also calculated to capture dynamic information, resulting in a 39-dimensional feature vector.
This document summarizes and compares different speech enhancement methods for noisy Punjabi speech at the phoneme level. It describes spectral subtraction, Wiener filtering, Kalman filtering, and RASTA methods. It also proposes a hybrid method using features from Wiener and Kalman filtering. Phonemes of Punjabi language and the methodology are explained. Results applying various noise types at different signal-to-noise ratios on phonemes show the proposed method produces the best enhancement compared to other methods.
The document proposes an objective method to assess speech quality in conversational contexts by taking into account talking quality, listening quality, and delay impact. It applies this approach to the results of four subjective tests on echo, delay, packet loss, and noise. A multiple linear regression is used to determine the relationship between conversational, talking, and listening qualities and delay. This achieves an accurate estimation of conversational scores with high correlation and low error between subjective and estimated scores. The relationship is then applied objectively by replacing subjective with objective talking and listening scores. The conversational model achieves high performance compared to existing standards.
This document summarizes a research paper on determining noise levels in noisy speech signals. It discusses extracting amplitude modulation spectrogram (AMS) features from noisy speech segments and classifying them as noise-dominated or signal-dominated using Bayes' classifier. The classified features are reconstructed using overlap-and-add to estimate the signal-to-noise ratio between the original clean speech and reconstructed speech, and between the original clean speech and noisy speech. The paper presents results on speech segments corrupted with different noises, showing SNR is reduced between the original clean and reconstructed speech compared to the original clean and noisy speech.
VAW-GAN for disentanglement and recomposition of emotional elements in speechKunZhou18
- The document describes a framework for emotional voice conversion using VAW-GAN that can disentangle and recompose emotional elements in speech. It proposes using VAW-GAN with continuous wavelet transform to model prosody and decompose fundamental frequency into different time scales. Conditioning the decoder on fundamental frequency is shown to improve emotion conversion performance. Experiments demonstrate the effectiveness of the approach on an English emotional speech database.
Review Paper on Noise Reduction Using Different TechniquesIRJET Journal
This document reviews different techniques for noise reduction in hearing aids. It discusses how digital hearing aids use algorithms like fast Fourier transforms and adaptive filters to separate speech from noise and improve speech comprehension in noisy environments. The document also evaluates different noise reduction algorithms and filter bank designs that have been proposed in previous research to enhance the noise reduction capabilities of hearing aids. The goal of developing improved noise reduction for hearing aids is to help hearing-impaired individuals better understand speech in various real-world settings with background noise.
This is the presentation of our IEEE ICASSP 2021 paper "seen and unseen emotional style transfer for voice conversion with a new emotional speech dataset".
The document summarizes Kun Zhou's PhD research on emotional voice conversion with non-parallel data at the National University of Singapore. It introduces emotional voice conversion and its challenges, including the lack of parallel training data. It then summarizes Kun's publications, which propose CycleGAN-based and VAW-GAN approaches to model prosody for speaker-dependent and independent emotional voice conversion. One publication introduces a method for transferring both seen and unseen emotional styles using a pre-trained speech emotion recognizer to describe emotional styles.
Voice recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech waves. This technique makes it possible to use the speaker's voice to verify their identity and control access to services such as voice dialing, banking by telephone, telephone shopping, database access services, information services, voice mail, security control for confidential information areas, and remote access to computers.
This document describes how to build a simple, yet complete and representative automatic speaker recognition system. Such a speaker recognition system has potential in many security applications. For example, users have to speak a PIN (Personal Identification Number) in order to gain access to the laboratory door, or users have to speak their credit card number over the telephone line to verify their identity. By checking the voice characteristics of the input utterance, using an automatic speaker recognition system similar to the one that we will describe, the system is able to add an extra level of security.
A Novel Uncertainty Parameter SR ( Signal to Residual Spectrum Ratio ) Evalua...sipij
This document presents a novel speech enhancement evaluation approach called SR (Signal to Residual spectrum ratio). SR aims to improve speech intelligibility for hearing impaired individuals in non-stationary noisy environments. The approach segments noisy speech into pure, quasi, and non-speech frames using threshold conditions on the signal and estimated noise spectra. Noise power is estimated differently for each frame type. SR and LLR (log likelihood ratio) are used to measure distortions and compare the proposed approach to weighted averaging techniques. Results show the proposed SR approach achieves better segmental SNR and LLR scores than weighted averaging, indicating it enhances speech quality and intelligibility more effectively in car, airport, and train noise environments.
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.
This document analyzes speech coding algorithms for Hindi and English languages. It discusses Linear Predictive Coding (LPC), an algorithm that accurately estimates speech parameters and represents speech signals at reduced bit rates while preserving quality. The paper proposes a voice-excited LPC algorithm and implements it on Hindi and English male and female voices. It analyzes tradeoffs between bit rates, delay, signal-to-noise ratio, and complexity. The results show low bit-rates and better signal-to-noise ratio with this algorithm.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Performance enhancement of dct based speaker recognition using wavelet de noi...eSAT Journals
Abstract Presence of noise in the speech signal is one of the major problems in Speaker Recognition. The speaker recognition performance gradually degrades as the intensity of noise increases. The system gives high accuracy when the speech signal is noise free, but in real life scenario getting a noise free speech signal is challenging. Hence, elimination of the noise from speech signal is an important aspect in speaker recognition process. This work uses wavelet based denoising of the recorded speech signal in order to enhance the performance of speaker recognition. In this paper, wavelet based denoising technique has been applied to the DCT based speaker recognition system which was proposed in our previous work. Additive white Gaussian noise has been added to the speech signal and performance analysis of the system has been done using different SNR value. Keywords: Wavelet denoising, AWGN, Speaker Recognition, Thresholding, DCT, Feature Extraction.
The document discusses speech processing using MATLAB. It involves analyzing speech sounds through components like normalizers and segmenters, converting the sounds to a code, and then synthesizing speech by reproducing recorded elements. There are many applications ranging from limited to full communication systems. Speech signals are usually processed digitally as a form of digital signal processing. Aspects include acquiring, manipulating, storing, transferring and outputting speech signals.
Voice Identification And Recognition System, MatlabSohaib Tallat
A simple yet complex approach to modern sophistication.
Made this project using the MFCC approach and then embedding the code to a Graphical User Interface. In the end made a standalone application for the program using deployment tools of matlab
El documento describe las características del Humanismo y el Renacimiento. Ambos movimientos se caracterizan por una visión antropocéntrica y recogen la herencia de la antigüedad grecorromana. Factores como la llegada de sabios bizantinos a Europa, la invención de la imprenta y el apoyo de mecenas financiaron el desarrollo del Humanismo y el Renacimiento.
O documento fornece dicas para curtir um festival de música de forma confortável e segura. Recomenda-se usar roupas leves e confortáveis, tênis em vez de chinelos, levar documentos, óculos de sol, celular, dinheiro e proteção solar. Também aconselha chegar cedo para ficar mais perto do palco e ter paciência na saída para evitar aglomerações.
Mer än 90 % av verksamheterna i Davidshallsområdet i Malmö, är privatägda. Inga större butiks- och restaurangkedjor finns vid torget. Alla är helt unika och tillhandahåller oftast varor handplockade för kunderna. Detta åskådliggörs med fotoutställningen ”Vi är Davidshallare” och lanseringen av områdets logotyp under Höstfesten 24/9.
Este documento propone varias medidas para garantizar el derecho a la vivienda desde un ayuntamiento de izquierdas, incluyendo potenciar una oficina antidesahucios, crear comisiones antidesahucio en los barrios, modificar impuestos para que las entidades financieras paguen en caso de dación en pago, evitar desahucios en viviendas públicas, crear un censo de viviendas vacías y promover ayudas al alquiler.
El documento explica qué es el reciclaje, por qué es importante y cómo funciona el proceso de separación de residuos. El reciclaje permite reutilizar materiales después de que un artículo ha llegado al final de su vida útil, lo que reduce los costos de eliminación de basura y ahorra recursos naturales como el petróleo y los árboles. Para reciclar correctamente, los residuos se deben separar en los contenedores adecuados según su tipo, como vidrio, papel, plástico o latas.
This document provides an overview of speech recognition systems and recent progress in the field. It discusses different types of speech recognition including isolated word, connected word, continuous speech, and spontaneous speech. Various techniques used in speech recognition are also summarized, such as simulated evolutionary computation, artificial neural networks, fuzzy logic, Kalman filters, and Hidden Markov Models. The document reviews several papers published between 2004-2012 that studied speech recognition methods including using dynamic spectral subband centroids, Kalman filters, biomimetic computing techniques, noise estimation, and modulation filtering. It concludes that Hidden Markov Models combined with MFCC features provide good recognition results and are suitable for large vocabulary, speaker-independent, continuous speech recognition.
A GAUSSIAN MIXTURE MODEL BASED SPEECH RECOGNITION SYSTEM USING MATLABsipij
1) The document describes the development of a speaker-dependent speech recognition system using MATLAB. It uses Gaussian mixture models for acoustic modeling and mel-frequency cepstral coefficients for feature extraction.
2) The system is designed to recognize isolated digits 0-9. Voice activity detection is performed to detect segments of speech. Various windowing functions are evaluated to reduce spectral leakage during feature extraction.
3) 13 MFCCs plus energy are extracted from each 30ms frame with 10ms shift. The first and second derivatives are also calculated to capture dynamic information, resulting in a 39-dimensional feature vector.
This document summarizes and compares different speech enhancement methods for noisy Punjabi speech at the phoneme level. It describes spectral subtraction, Wiener filtering, Kalman filtering, and RASTA methods. It also proposes a hybrid method using features from Wiener and Kalman filtering. Phonemes of Punjabi language and the methodology are explained. Results applying various noise types at different signal-to-noise ratios on phonemes show the proposed method produces the best enhancement compared to other methods.
The document proposes an objective method to assess speech quality in conversational contexts by taking into account talking quality, listening quality, and delay impact. It applies this approach to the results of four subjective tests on echo, delay, packet loss, and noise. A multiple linear regression is used to determine the relationship between conversational, talking, and listening qualities and delay. This achieves an accurate estimation of conversational scores with high correlation and low error between subjective and estimated scores. The relationship is then applied objectively by replacing subjective with objective talking and listening scores. The conversational model achieves high performance compared to existing standards.
This document summarizes a research paper on determining noise levels in noisy speech signals. It discusses extracting amplitude modulation spectrogram (AMS) features from noisy speech segments and classifying them as noise-dominated or signal-dominated using Bayes' classifier. The classified features are reconstructed using overlap-and-add to estimate the signal-to-noise ratio between the original clean speech and reconstructed speech, and between the original clean speech and noisy speech. The paper presents results on speech segments corrupted with different noises, showing SNR is reduced between the original clean and reconstructed speech compared to the original clean and noisy speech.
VAW-GAN for disentanglement and recomposition of emotional elements in speechKunZhou18
- The document describes a framework for emotional voice conversion using VAW-GAN that can disentangle and recompose emotional elements in speech. It proposes using VAW-GAN with continuous wavelet transform to model prosody and decompose fundamental frequency into different time scales. Conditioning the decoder on fundamental frequency is shown to improve emotion conversion performance. Experiments demonstrate the effectiveness of the approach on an English emotional speech database.
Review Paper on Noise Reduction Using Different TechniquesIRJET Journal
This document reviews different techniques for noise reduction in hearing aids. It discusses how digital hearing aids use algorithms like fast Fourier transforms and adaptive filters to separate speech from noise and improve speech comprehension in noisy environments. The document also evaluates different noise reduction algorithms and filter bank designs that have been proposed in previous research to enhance the noise reduction capabilities of hearing aids. The goal of developing improved noise reduction for hearing aids is to help hearing-impaired individuals better understand speech in various real-world settings with background noise.
This is the presentation of our IEEE ICASSP 2021 paper "seen and unseen emotional style transfer for voice conversion with a new emotional speech dataset".
The document summarizes Kun Zhou's PhD research on emotional voice conversion with non-parallel data at the National University of Singapore. It introduces emotional voice conversion and its challenges, including the lack of parallel training data. It then summarizes Kun's publications, which propose CycleGAN-based and VAW-GAN approaches to model prosody for speaker-dependent and independent emotional voice conversion. One publication introduces a method for transferring both seen and unseen emotional styles using a pre-trained speech emotion recognizer to describe emotional styles.
Voice recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech waves. This technique makes it possible to use the speaker's voice to verify their identity and control access to services such as voice dialing, banking by telephone, telephone shopping, database access services, information services, voice mail, security control for confidential information areas, and remote access to computers.
This document describes how to build a simple, yet complete and representative automatic speaker recognition system. Such a speaker recognition system has potential in many security applications. For example, users have to speak a PIN (Personal Identification Number) in order to gain access to the laboratory door, or users have to speak their credit card number over the telephone line to verify their identity. By checking the voice characteristics of the input utterance, using an automatic speaker recognition system similar to the one that we will describe, the system is able to add an extra level of security.
A Novel Uncertainty Parameter SR ( Signal to Residual Spectrum Ratio ) Evalua...sipij
This document presents a novel speech enhancement evaluation approach called SR (Signal to Residual spectrum ratio). SR aims to improve speech intelligibility for hearing impaired individuals in non-stationary noisy environments. The approach segments noisy speech into pure, quasi, and non-speech frames using threshold conditions on the signal and estimated noise spectra. Noise power is estimated differently for each frame type. SR and LLR (log likelihood ratio) are used to measure distortions and compare the proposed approach to weighted averaging techniques. Results show the proposed SR approach achieves better segmental SNR and LLR scores than weighted averaging, indicating it enhances speech quality and intelligibility more effectively in car, airport, and train noise environments.
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.
This document analyzes speech coding algorithms for Hindi and English languages. It discusses Linear Predictive Coding (LPC), an algorithm that accurately estimates speech parameters and represents speech signals at reduced bit rates while preserving quality. The paper proposes a voice-excited LPC algorithm and implements it on Hindi and English male and female voices. It analyzes tradeoffs between bit rates, delay, signal-to-noise ratio, and complexity. The results show low bit-rates and better signal-to-noise ratio with this algorithm.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Performance enhancement of dct based speaker recognition using wavelet de noi...eSAT Journals
Abstract Presence of noise in the speech signal is one of the major problems in Speaker Recognition. The speaker recognition performance gradually degrades as the intensity of noise increases. The system gives high accuracy when the speech signal is noise free, but in real life scenario getting a noise free speech signal is challenging. Hence, elimination of the noise from speech signal is an important aspect in speaker recognition process. This work uses wavelet based denoising of the recorded speech signal in order to enhance the performance of speaker recognition. In this paper, wavelet based denoising technique has been applied to the DCT based speaker recognition system which was proposed in our previous work. Additive white Gaussian noise has been added to the speech signal and performance analysis of the system has been done using different SNR value. Keywords: Wavelet denoising, AWGN, Speaker Recognition, Thresholding, DCT, Feature Extraction.
The document discusses speech processing using MATLAB. It involves analyzing speech sounds through components like normalizers and segmenters, converting the sounds to a code, and then synthesizing speech by reproducing recorded elements. There are many applications ranging from limited to full communication systems. Speech signals are usually processed digitally as a form of digital signal processing. Aspects include acquiring, manipulating, storing, transferring and outputting speech signals.
Voice Identification And Recognition System, MatlabSohaib Tallat
A simple yet complex approach to modern sophistication.
Made this project using the MFCC approach and then embedding the code to a Graphical User Interface. In the end made a standalone application for the program using deployment tools of matlab
El documento describe las características del Humanismo y el Renacimiento. Ambos movimientos se caracterizan por una visión antropocéntrica y recogen la herencia de la antigüedad grecorromana. Factores como la llegada de sabios bizantinos a Europa, la invención de la imprenta y el apoyo de mecenas financiaron el desarrollo del Humanismo y el Renacimiento.
O documento fornece dicas para curtir um festival de música de forma confortável e segura. Recomenda-se usar roupas leves e confortáveis, tênis em vez de chinelos, levar documentos, óculos de sol, celular, dinheiro e proteção solar. Também aconselha chegar cedo para ficar mais perto do palco e ter paciência na saída para evitar aglomerações.
Mer än 90 % av verksamheterna i Davidshallsområdet i Malmö, är privatägda. Inga större butiks- och restaurangkedjor finns vid torget. Alla är helt unika och tillhandahåller oftast varor handplockade för kunderna. Detta åskådliggörs med fotoutställningen ”Vi är Davidshallare” och lanseringen av områdets logotyp under Höstfesten 24/9.
Este documento propone varias medidas para garantizar el derecho a la vivienda desde un ayuntamiento de izquierdas, incluyendo potenciar una oficina antidesahucios, crear comisiones antidesahucio en los barrios, modificar impuestos para que las entidades financieras paguen en caso de dación en pago, evitar desahucios en viviendas públicas, crear un censo de viviendas vacías y promover ayudas al alquiler.
El documento explica qué es el reciclaje, por qué es importante y cómo funciona el proceso de separación de residuos. El reciclaje permite reutilizar materiales después de que un artículo ha llegado al final de su vida útil, lo que reduce los costos de eliminación de basura y ahorra recursos naturales como el petróleo y los árboles. Para reciclar correctamente, los residuos se deben separar en los contenedores adecuados según su tipo, como vidrio, papel, plástico o latas.
El documento describe el modelo del sistema solar, incluyendo que está compuesto por el Sol y los planetas que giran alrededor de él, y las leyes que rigen los movimientos, como las leyes de Kepler y la ley de la gravedad de Newton. Explica las teorías sobre cómo se formó el sistema solar a partir de una nebulosa primitiva que se fue condensando bajo la influencia de la gravedad.
Lightning Talk #9: How UX and Data Storytelling Can Shape Policy by Mika Aldabaux singapore
How can we take UX and Data Storytelling out of the tech context and use them to change the way government behaves?
Showcasing the truth is the highest goal of data storytelling. Because the design of a chart can affect the interpretation of data in a major way, one must wield visual tools with care and deliberation. Using quantitative facts to evoke an emotional response is best achieved with the combination of UX and data storytelling.
This document summarizes a study of CEO succession events among the largest 100 U.S. corporations between 2005-2015. The study analyzed executives who were passed over for the CEO role ("succession losers") and their subsequent careers. It found that 74% of passed over executives left their companies, with 30% eventually becoming CEOs elsewhere. However, companies led by succession losers saw average stock price declines of 13% over 3 years, compared to gains for companies whose CEO selections remained unchanged. The findings suggest that boards generally identify the most qualified CEO candidates, though differences between internal and external hires complicate comparisons.
This document discusses determining noise levels in noisy speech signals. It presents the following:
1) Amplitude modulation spectrogram (AMS) features are extracted from noisy speech segments and classified as noise-dominated or signal-dominated using Bayes' classifier.
2) The classified features are reconstructed using overlap-and-add to obtain a reconstructed speech segment.
3) The signal-to-noise ratio (SNR) is calculated between the reconstructed speech segment and the original noises, as well as between the clean speech signal and noisy speech segment.
Speech Enhancement for Nonstationary Noise Environmentssipij
In this paper, we present a simultaneous detection and estimation approach for speech enhancement in nonstationary noise environments. A detector for speech presence in the short-time Fourier transform domain is combined with an estimator, which jointly minimizes a cost function that takes into account both detection and estimation errors. Under speech-presence, the cost is proportional to a quadratic spectral amplitude error, while under speech-absence, the distortion depends on a certain attenuation factor. Experimental results demonstrate the advantage of using the proposed simultaneous detection and estimation approach which facilitate suppression of nonstationary noise with a controlled level of speech distortion.
This document summarizes a research paper on speech enhancement using the signal subspace algorithm. It begins with an abstract describing how noise degrades speech quality and intelligibility in communication systems. It then provides background on speech enhancement objectives and commonly used methods like spectral subtraction and signal subspace. The paper describes the signal subspace algorithm and shows its ability to enhance speech signals by suppressing noise. Experimental results on sine waves with added Gaussian noise demonstrate improved peak signal-to-noise ratios when using the signal subspace method compared to the noisy signals. The conclusion is that the algorithm removes noise to a great extent from noisy speech.
Teager Energy Operation on Wavelet Packet Coefficients for Enhancing Noisy Sp...CSCJournals
In this paper a new thresholding based speech enhancement approach is presented, where the threshold is statistically determined by employing the Teager energy operation on the Wavelet Packet (WP) coefficients of noisy speech. The threshold thus obtained is applied on the WP coefficients of the noisy speech by using a hard thresholding function in order to obtain an enhanced speech. Detailed simulations are carried out in the presence of white, car, pink, and babble noises to evaluate the performance of the proposed method. Standard objective measures, spectrogram representations and subjective listening tests show that the proposed method outperforms the existing state-of-the-art thresholding based speech enhancement approaches for noisy speech from high to low levels of SNR.
This document summarizes a research paper on spectral analysis techniques for speech processing. It discusses how spectral subtraction is commonly used to reduce additive background noise in speech signals. The proposed method is a frequency-dependent spectral subtraction approach. Unlike conventional spectral subtraction that subtracts noise estimates uniformly across the entire spectrum, the proposed method allows flexible control over subtraction levels to reduce artifacts. Results show the method produces enhanced speech with lower residual noise and better quality than conventional spectral subtraction. In conclusion, a multi-band speech enhancement strategy is developed based on spectral subtraction to improve noise reduction while minimizing speech distortion.
Speech enhancement using spectral subtraction technique with minimized cross ...eSAT Journals
Abstract The aim of speech enhancement is to get significant reduction of noise and enhanced speech from noisy speech. There are several
approaches for speech enhancement .earlier approaches didn’t consider cross spectral terms into account. Cross spectral terms
become prominent when processing window size becomes small i.e. 20ms-30ms. In this paper, an enhancement method is
proposed for significant reduction of noise, and improvement in the quality and perceptibility of speech degraded by correlated
additive background noise. The proposed method is based on the spectral subtraction technique. The simple spectral subtraction
technique results in poor reduction of noise. One of the main reasons for this is neglecting the cross spectral terms of speech and
noise, based on the appropriation that clean speech and noise signals are completely uncorrelated to each other, which is not true
on short time basis. In this paper an improvement in reduction of the noise is achieved as compared to the earlier methods. This
fact is mainly attributed to the cross spectral terms between speech and noise. This algorithm can be implemented and used in
hearing aids for the benefit of hearing impaired people. Objective speech quality measures, spectrogram analyses and subjective
listening tests conforms the proposed method is more effective in comparison with earlier speech enhancement techniques.
Keywords: Spectral Subtaction,Cross Spectral Components
This document provides an overview of speech recognition systems and recent progress in the field. It discusses different types of speech recognition including isolated word, connected word, continuous speech, and spontaneous speech. Various techniques used in speech recognition are also summarized, such as simulated evolutionary computation, artificial neural networks, fuzzy logic, Kalman filters, and Hidden Markov Models. The document reviews several papers published between 2004-2012 that studied speech recognition methods including using dynamic spectral subband centroids, Kalman filters, biomimetic computing techniques, noise estimation, and modulation filtering. It concludes that Hidden Markov Models combined with MFCC features provide good recognition results for large vocabulary, speaker-independent, continuous speech recognition.
Single Channel Speech Enhancement using Wiener Filter and Compressive Sensing IJECEIAES
The speech enhancement algorithms are utilized to overcome multiple limitation factors in recent applications such as mobile phone and communication channel. The challenges focus on corrupted speech solution between noise reduction and signal distortion. We used a modified Wiener filter and compressive sensing (CS) to investigate and evaluate the improvement of speech quality. This new method adapted noise estimation and Wiener filter gain function in which to increase weight amplitude spectrum and improve mitigation of interested signals. The CS is then applied using the gradient projection for sparse reconstruction (GPSR) technique as a study system to empirically investigate the interactive effects of the corrupted noise and obtain better perceptual improvement aspects to listener fatigue with noiseless reduction conditions. The proposed algorithm shows an enhancement in testing performance evaluation of objective assessment tests outperform compared to other conventional algorithms at various noise type conditions of 0, 5, 10, 15 dB SNRs. Therefore, the proposed algorithm significantly achieved the speech quality improvement and efficiently obtained higher performance resulting in better noise reduction compare to other conventional algorithms.
This document summarizes a research paper that presents a speech enhancement method using stationary wavelet transform. The method first classifies speech into voiced, unvoiced, and silence regions based on short-time energy. It then applies different thresholding techniques to the wavelet coefficients of each region - modified hard thresholding for voiced speech, semi-soft thresholding for unvoiced speech, and setting coefficients to zero for silence. Experimental results using speech from the TIMIT database corrupted with white Gaussian noise at various SNR levels show improved performance over other popular denoising methods.
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.
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.
This document summarizes the key components of a voice recognition system, including signal modeling and pattern matching. Signal modeling represents converting speech signals into parameters through operations like spectral shaping and feature extraction. Feature extraction analyzes speech signals through temporal and spectral analysis techniques to obtain parameters like power, pitch, and vocal tract configuration. Pattern matching finds the parameter set from memory that most closely matches the input speech parameters. The document then discusses specific temporal analysis techniques like power and energy analysis, and spectral analysis techniques like filter banks, cepstral analysis, and linear predictive coding analysis used for feature extraction in voice recognition systems.
A REVIEW OF LPC METHODS FOR ENHANCEMENT OF SPEECH SIGNALSijiert bestjournal
This paper presents a review of LPC methods for enhancement of speech signals. The purpose of all method of speech enhancement is to impr ove quality of speech signal by minimizing the background noise . This paper especially comments on Power Spectral Subtraction,Multiband Spectral Subtraction,Non - Linear Spectral Subtraction Method,MMSE Spectral Subtraction Method and Spectral Subtraction based on perceptual properties. As the spectral subtraction produces the Residual noise,Musical noise,Linear Predictive Analysis method is used to enhance the Speech.
Enhanced modulation spectral subtraction incorporating various real time nois...IRJET Journal
This document summarizes a research paper on enhancing speech intelligibility using an enhanced modulation spectral subtraction (EMSS) method. The paper introduces background on speech enhancement and intelligibility. It describes the EMSS framework which applies spectral subtraction in the modulation domain using analysis-modification-synthesis. Experimental results on the NOIZEUS database show the EMSS method improves speech intelligibility measures like STOI and SNRseg over baseline methods for noises like car, airport and babble noise. The EMSS provides around 50-60% improvement in SNRseg over other modulation spectral subtraction methods. The paper concludes the EMSS is effective for speech enhancement applications in noisy environments like vehicles.
Speech Enhancement Based on Spectral Subtraction Involving Magnitude and Phas...IRJET Journal
This document presents a speech enhancement method based on spectral subtraction involving the magnitude and phase components. The proposed method aims to improve noisy speech quality by estimating and subtracting noise from the noisy speech signal in the frequency domain. It involves segmenting the noisy speech into overlapping frames, estimating the noise spectrum during non-speech periods, subtracting the noise magnitude from the noisy speech magnitude, and reconstructing the enhanced speech using the inverse FFT and overlap-add processing. The method was tested on different noise types using MATLAB simulations. Results showed the proposed method achieved better noise reduction compared to conventional spectral subtraction while introducing less speech distortion.
A Dual-Microphone Speech Enhancement Algorithm Based on the Coherence FunctionVikashSharma325767
This document proposes a dual-microphone speech enhancement algorithm based on the coherence function to improve speech intelligibility in noisy environments. It uses the coherence function between microphone signals to determine if a time-frequency region contains speech or noise. A filter is designed based on the real and imaginary parts of the coherence function to suppress noise while passing speech. The algorithm was implemented in MATLAB and shown to enhance speech intelligibility without requiring noise statistics, making it a potential method for future hearing aid devices.
Наше приложение (iOs, Android) и драйвер(Windows) представлены в независимом исследовании устройств для обучения студентов с ослабленным слухом.
The article by Saketh Sharma, Nitya Tiwari, and Prem C. Pandey in the "Int. Conf. on Intelligent Human Computer Interaction" proceedings (www.petralex.pro)
Implementation of a Digital Hearing Aid with User-Settable Frequency Response...IAMCP MENTORING
The article by Saketh Sharma, Nitya Tiwari, and Prem C. Pandey in the "Int. Conf. on Intelligent Human Computer Interaction" proceedings describes among others the Petralex smartphone App as a hearing aid - the product of the IAMCP members - IT4You (www.petralex.pro)
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Audio Noise Removal – The State of the Artijceronline
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.
1. Y Prasanthi, K Jyothi, GIET / International Journal of Engineering Research and Applications
(IJERA) ISSN: 2248-9622 www.ijera.com
Vol. 2, Issue4, July-August 2012, pp.2162-2165
Frequency Depended Spectral Subtraction For Speech
Enhancement
Y Prasanthi, K Jyothi, GIET
Abstract:
The corruption of speech due to presence of To understand what the speaker is saying. Noise
additive background noise causes severe difficulties in reduction or speech enhancement algorithms are used to
various communications environments. This paper suppress such background noise and improve the
addresses the problem of reduction of additive perceptual quality and intelligibility of speech. Even
background noise in speech. The proposed approach is though speech is perceptible in a moderately noisy
a frequency dependent speech enhancement method environment, many applications like mobile
based on the proven spectral subtraction method. Most communications, speech recognition and aids for the
implementations and variations of the basic spectral hearing handicapped, to name a few, drive the effort to
subtraction technique advocate subtraction of the noise build more effective noise reduction algorithms for
spectrum estimate over the entire speech spectrum. better performance. Over the years engineers have
However real world noise is mostly colored and does developed a variety of theoretical and relatively
not affect the speech signal uniformly over the entire effective techniques to combat this issue. However, the
spectrum. In this paper, new spectral subtraction problem of cleaning noisy speech still poses a challenge
method proposed which takes into account the fact that to the area of signal processing. Removing various
colored noise affects the speech spectrum differently at types of noise is difficult due to the random nature of
various frequencies. This method provides a greater the noise and the inherent complexities of speech.
degree of flexibility and control on the noise subtraction Noise reduction techniques usually have a trade off
levels that reduces artifacts in the enhanced speech, between the amount of noise removal and speech
resulting in improved speech quality. This method distortions introduced due the processing of the speech
outperforms the conventional spectral subtraction signal. Complexity and ease of implementation of the
method with respect to speech quality and reduced noise reduction algorithms is also of concern in
musical noise. applications especially those related to portable devices
such as mobile communications and digital hearing
1. INTRODUCTION aids. The spectral subtraction method is a well-known
A major part of the interaction between noise reduction technique. Most implementations and
humans takes place via speech communication. variations of the basic technique advocate subtraction
Hence, research in speech and hearing sciences has of the noise spectrum estimate over the entire speech
been going on for centuries to understand the dynamics spectrum. However, real world noise is mostly colored
and processes involved in the production and and does not affect the speech signal uniformly over the
perception of speech. The field of speech processing is entire spectrum. In this paper, we propose a multi-band
essentially an application of signal processing spectral subtraction approach that takes into account the
techniques to acoustic signals using the knowledge fact that colored noise affects the speech spectrum
offered by researchers in the field of hearing sciences. differently at various frequencies. This method
The explosive advances in recent years in the field of outperforms the standard power spectral subtraction
digital computing have provided a tremendous boost to method resulting in superior speech quality and largely
the field of speech processing. Digital signal processing reduced musical noise.
techniques are more sophisticated and advanced as
compared to their analog counterparts. Ease and speed 1.2 Short-term Spectral Amplitude Techniques
of representing, storing, retrieving and processing The short-term spectral amplitude (STSA) of speech
speech data has contributed to the development of has been exploited successfully in the development of
efficient and effective speech processing techniques to various speech enhancement algorithms. The basic idea
address the issues related to speech. The presence of is to use the STSA of the noisy speech input and
background noise in speech significantly reduces the recover an estimate of the clean STSA by removing the
intelligibility of speech. Degradation of speech severely part contributed by the additive noise. Expressed as
affects the ability of person, whether impaired or
normal hearing,
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2. Y Prasanthi, K Jyothi, GIET / International Journal of Engineering Research and Applications
(IJERA) ISSN: 2248-9622 www.ijera.com
Vol. 2, Issue4, July-August 2012, pp.2162-2165
values in the enhanced spectrum were floored to the
noisy spectrum as
(1)
Where S(k) and Y(k) are magnitude spectrum of clean
speech and the noisy speech. Estimated noise
magnitude is calculated during periods of
speech, α is an over-subtraction factor. (3)
A general representation of STSA technique is given in
Figure 1. Where The spectral floor parameter was set to β=0.001.
3. IMPLEMENTATION AND
PERFORMANCE EVALUTION
3.1 Objective Measures for Performance Evaluation
In the evaluation of speech enhancement
algorithm, it is necessary to identify the similarities and
differences in perceived quality and subjectively
measured intelligibility. Speech quality is an indicator
of the naturalness of the processed speech signal.
Intelligibility of speech signals is a measure of the
amount of speech information present in the signal that
is responsible for conveying what that speaker is
saying. The interrelationship between perceived speech
Figure 1: Diagrammatic Representation of the short- and intelligibility is not clearly understood.
time Spectral Magnitude Enhancement System Performance evaluation tests can be done by subjective
quality measures or objective quality measures. While
The input to the system is the noise-corrupted subjective measures provide a broad measure of
signal y(n) . While there are many methods for the performance since a large difference in quality is
analysis-synthesis processing, the Short-term Fourier necessary to be distinguishable to the listener. Hence, it
Transform (STFT) of the signal with OverLap and Add becomes difficult to get a reliable measure of changes
(OLA) is the most commonly used method. Spectral due to algorithm parameters. Objective measures, on
subtraction is a well-known noise reduction method the other hand, provide a measure that can be easily
based on the STSA estimation technique. The basic implemented and reliably reproduced.
power spectral subtraction technique, as proposed by It is necessary to conduct off-line simulations
Boll, is popular due to its simple underlying concept to check the validity and feasibility of an algorithm
and its effectiveness in enhancing speech degraded by before it can be implemented on a real-time system.
additive noise but drawback of the power spectral Implementation on a workstation permits modifications
subtraction technique is residual noise. and changes to the algorithm without constraints of
time, memory or computational power. The simulations
2. FREQUENCY DEPENDED SPECTRAL were carried out on an using Matlab, a technical
SUBTRACTION computing software. The speech signal is first
Along with the actual noise suppression Hamming windowed using a 20-ms window and a 10-
operation, some pre-processing methods are needed to ms overlap between frames. The windowed speech
achieving good speech quality. In frequency depended frame is then analyzed using the Fast Fourier Transform
spectral subtraction, the speech spectrum is divided into (FFT). Smoothing of the magnitude spectrum was
non-overlapping bands, and spectral subtraction is found to reduce the variance of the speech spectrum
performed independently in each band. The fact that and contribute to the enhancement in speech quality. A
colored noise affects the speech spectrum differently at weighted spectral average is taken over preceding and
various frequencies. succeeding frames of data as given by Equation.
(4)
(2) Where j is the frame index 0<Wi<1. The averaging is
Where bi and ei are beginning and ending done over M preceding and succeeding frames of
frequency bins of the ith frequency band.δi is tweaking speech. The number of frames M is limited to 2 to
factor that can be individually set for each band to prevent smearing of the speech spectral content. The
customize the noise removal properties.The negative
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3. Y Prasanthi, K Jyothi, GIET / International Journal of Engineering Research and Applications
(IJERA) ISSN: 2248-9622 www.ijera.com
Vol. 2, Issue4, July-August 2012, pp.2162-2165
filter weights Wi were empirically determined and set
to W= [0.09, 0.25, 0.32, 0.25, 0.09]
(5)
Where fi is the upper frequency of the ith
band, and Fs is the sampling frequency in Hz. The Figure 2: Intelligibility Test Results for Seven Subjects
motivation for using smaller δi values for the low scored on Percentage Words Correct
frequency bands is to minimize speech distortion, since
most of the speech energy is present in the lower
frequencies. Relaxed subtraction is also used for the
high frequency bands. Subtraction is performed over
each band as indicated in equation and the negative
values are rectified using the spectral floor. A small
amount of the original noisy spectrum can be
introduced back into the enhanced spectrum to mask
any remaining musical noise. In this implementation,
5% of the original noisy spectrum within each band is
combined, and the enhanced signal is obtained by
taking the IFFT of the enhanced spectrum using the Figure 3: signal representation of the sentence ''I am
phase of the original noisy spectrum. Finally, the David''. The top representation is the original signal, the
standard overlap-and-add method is used to obtain the middle representation is corrupted signal, and the
enhanced signal. bottom representation is the enhanced signal obtained
by the frequency depended spectral subtraction
3.2 Subjective Evaluation of Speech Intelligibility method
Subjective tests are conducted by having some
human subjects listen to the prepared test speech files
and evaluate based on some criteria. Intelligibility test
were carried out at the Callier Center for
Communication Disorders/UTD on seven hearing-
impaired subjects with severe to profound hearing loss.
Speech enhanced by the MBSS algorithm with four
linearly spaced frequency bands was evaluated against
the noisy speech. The sentences were corrupted using
speech-shaped noise at 0 dB SNR. The noise-corrupted
sentences were played in a random order through Figure 4: Spectrogram of the sentence ''I am David''.
speakers in a sound insulated booth. The subjects were The top spectrogram is the original signal, the middle
asked to repeat the sentence they heard. Intelligibility spectrogram is the corrupted signal, the bottom
was measured in terms of percentage of words correct. spectrogram is the enhanced signal obtained by
Figure 2 gives the bar plots of the intelligibility scores frequency depended spectral subtraction method.
achieved by each subject and the average score for both
the test conditions. The results obtained by objective
evaluation are an indicator of the best speech quality 5. CONCLUSIONS
that can be obtained by the different configurations of The work in this paper addressed the problem
the algorithm. The proposed multi-band spectral of enhancing speech in noisy conditions. A multi-band
subtraction approach(with number of bands>3) spectral subtraction method, based on the direct
performed better than the PSS approach for both SNRs. estimation of the short-term spectral amplitude of
speech and the non-uniform effect of noise on speech.
4. RESULTS The results establish the superiority of the proposed
method over the conventional spectral subtraction
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4. Y Prasanthi, K Jyothi, GIET / International Journal of Engineering Research and Applications
(IJERA) ISSN: 2248-9622 www.ijera.com
Vol. 2, Issue4, July-August 2012, pp.2162-2165
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