scholarly journals HIGH QUALITY LOW BITRATE VOICE CODEC FOR TRANSMISSION OVER ADVANCED LTE

2021 ◽  
Vol 1 (1) ◽  
pp. 83-93
Author(s):  
Noor N. Edan ◽  
Nasser N. Khamiss
Keyword(s):  
Communication Systems ◽  
Linear Prediction ◽  
Voice Quality ◽  
Open Loop ◽  
Spatial Multiplexing ◽  
Bandwidth Utilization ◽  
Available Bandwidth ◽  
Voice Signal ◽  
Channel Bandwidth ◽  
The Voice

In mobile communication systems bit-rate reductions while maintaining an acceptable voice quality are necessary to achieve efficiency in channel bandwidth utilization and users satisfaction. As Long-Term Evolution(LTE) converging towards all-IP solutions and supporting VOIP service, the voice signals are converted into coded digital bit-stream and sent over the network. This paper proposes the implementation of codebook excited linear prediction (CELP) voice codec algorithm based on two source-rates of low 9.6Kbps and medium 16Kbps for achieving a perceptible level of voice quality, while efficiently using available bandwidth during the transmission over advanced LTE. The architecture of proposed CELP codec model is implemented to decompose the voice signal into a set of parameters that characterize each particular frame at the encoder part, these parameters are quantized and encoded for transmission to the decoder. The investigation showed that the configuration of the link and the applied CELP codec mode mainly influence on the obtained voice capacity and quality. The quantifying also shows that the voice quality can be traded for the enhanced capacity, since the low rate codec will produce lower voice quality than higher rate codec. Also, this paper is achieved, during theconfiguration of the system with higher channel quality indicator (CQI) index, increasing in the capacity gain to a saturated value of about 500 and 1000 users per cell over 5MHz bandwidth for transmit diversity (TD) and Open-Loop Spatial Multiplexing (OLSM) respectively and up to 1000 and 2000 users per cell over 10MHz channel bandwidth for TD and OLSM respectively.

scholarly journals ERIL: An Algorithm for Emotion Recognition From Indian Languages Using Machine Learning

2021 ◽  
Author(s):  
PRAMOD MEHRA ◽  
Parag Jain
Keyword(s):  
Emotion Recognition ◽  
Linear Prediction ◽  
Voice Quality ◽  
Human Interaction ◽  
Indian Languages ◽  
Mel Frequency Cepstral Coefficients ◽  
Voice Signal ◽  
Proposed Model ◽  
Human Ear ◽  
The Voice

Abstract For a human interaction with machine, it is important that it understand the mood of the speaker. Until now we train machines on neutral speeches or utterances. The mood of a person would affect their performances. Deciphering human mood is challenging for the machines, as human can create fourteen distinct sound in a second. For a machine to understand the human behaviour, it should understand the acoustic abilities of the human ear. Mel Frequency Cepstral Coefficients (MFCC) and Linear Prediction coefficients (LPC) can replicate human auditory system. The proposed model Emotion Recognition from Indian Languages (ERIL) extracts emotions like fear, anger, surprise, sadness, happiness, and neutral. ERIL first pre-processes the voice signal, extracts selective MFCC, LPC, pitch, and voice quality features, then classifies the speech using Catboost. ERIL is a multilingual emotion classifier, it is independent of any language. We checked it on Hindi, Gujarati, Marathi, Punjabi, Bangla, Tamil, Oriya, and Telugu. We recorded a speech dataset of various emotions in these languages. ERIL is compared to other benchmark classifiers.

scholarly journals Voice re-education and its evaluation through digital signal processing: A case study

Loquens ◽  
2017 ◽  
Vol 4 (1) ◽  
pp. 040
Author(s):  
Zulema Santana-López ◽  
Óscar Domínguez-Jaén ◽  
Jesús B. Alonso ◽  
María Del Carmen Mato-Carrodeguas
Keyword(s):  
Voice Quality ◽  
Quality Measurement ◽  
Digital Signal ◽  
Communication Process ◽  
Time Frequency ◽  
Voice Signal ◽  
Functional Dysphonia ◽  
Before And After ◽  
Voice Pathologies ◽  
The Voice

Voice pathologies, caused either by functional dysphonia or organic lesions, or even by just an inappropriate emission of the voice, may lead to vocal abuse, affecting significantly the communication process. The present study is based on the case of a single patient diagnosed with myasthenia gravis (Erb-Goldflam syndrome). In this case, this affection has caused, among other disruptions, a dysarthria. For its treatment, a technique for the education and re-education of the voice has been used, based on a resonator element: the cellophane screen. This article shows the results obtained in the patient after applying a vocal re-education technique called the Cimardi Method: the Cellophane Screen, which is a pioneering technique in this field. Changes in the patient’s voice signal have been studied before and after the application of the Cimardi Method in different domains of study: time-frequency, spectrum, and cepstrum. Moreover, parameters for voice quality measurement, such as shimmer, jitter and harmonic-to-noise ratio (HNR), have been used to quantify the results obtained with the Cimardi Method. Once the results were analyzed, it has been observed that the Cimardi Method helps to produce a more natural and free vocal emission, which is very useful as a rehabilitation therapy for those people presenting certain vocal disorders.

A Methodological Study of Perturbation and Additive Noise in Synthetically Generated Voice Signals

1987 ◽  
Vol 30 (4) ◽  
pp. 448-461 ◽  
Author(s):  
James Hillenbrand
Keyword(s):  
Additive Noise ◽  
Signal To Noise Ratio ◽  
Measurement Techniques ◽  
Voice Quality ◽  
Large Body ◽  
General Purpose ◽  
Measured Signal ◽  
Acoustic Measures ◽  
Voice Signal ◽  
The Voice

There is a relatively large body of research that is aimed at finding a set of acoustic measures of voice signals that can be used to: (a) aid in the detection, diagnosis, and evaluation of voice-quality disorders; (b) identify individual speakers by their voice characteristics; or (c) improve methods of voice synthesis. Three acoustic parameters that have received a relatively large share of attention, especially in the voice-disorders literature, are pitch perturbation, amplitude perturbation, and additive noise. The present study consisted of a series of simulations using a general-purpose formant synthesizer that were designed primarily to determine whether these three parameters could be measured independent of one another. Results suggested that changes in any single dimension can affect measured values of all three parameters. For example, adding noise to a voice signal resulted not only in a change in measured signal-to-noise ratio, but also in measured values of pitch and amplitude perturbation, These interactions were quite large in some cases, especially in view of the fact that the perturbation phenomena that are being measured are generally quite small. For the most part, the interactions appear to be readily explainable when the measurement techniques are viewed in relation to what is known about the acoustics of voice production.

SYNCHRONIZATION OF LORENZ SYSTEMS VIA IMPULSIVE CONTROL AND SPORADIC COUPLING

2004 ◽  
Vol 14 (08) ◽  
pp. 2963-2969 ◽  
Author(s):  
K. LI ◽  
Y. C. SOH ◽  
Z. G. LI
Keyword(s):  
Secure Communication ◽  
Communication Systems ◽  
Impulsive Control ◽  
Time Interval ◽  
Secure Communications ◽  
Bandwidth Utilization ◽  
Channel Bandwidth ◽  
Time Required

In this paper, a method that combines impulsive control and sporadic coupling for the synchronization of two Lorenz systems is proposed. The time required to synchronize the Lorenz systems can be readily computed. With this scheme, the Lorenz systems can be synchronized quickly and the time interval between successive impulsive signals can be made larger. This will improve the efficiency of channel bandwidth utilization when applied to secure communications. The results obtained in this paper are very useful in the applications of chaos-based secure communication systems.

scholarly journals A System View on Iterative MIMO Detection: Dynamic Sphere Detection versus Fixed Effort List Detection

VLSI Design ◽  
2012 ◽  
Vol 2012 ◽  
pp. 1-14 ◽  
Author(s):  
Christina Gimmler-Dumont ◽  
Frank Kienle ◽  
Bin Wu ◽  
Guido Masera
Keyword(s):  
Communication Systems ◽  
Closed Loop ◽  
Mimo Systems ◽  
Open Loop ◽  
Spatial Multiplexing ◽  
Loop System ◽  
Mimo Detection ◽  
Closed Loop System ◽  
Closed Loop Systems ◽  
Sphere Detection

Multiple-antenna systems are a promising approach to increase the data rate of wireless communication systems. One efficient possibility is spatial multiplexing of the transmitted symbols over several antennas. Many different MIMO detector algorithms exist for this spatial multiplexing. The major difference between different MIMO detectors is the resulting communications performance and implementation complexity, respectively. Particularly closed-loop MIMO systems have attained a lot of attention in the last years. In a closed-loop system, reliability information is fed back from the channel decoder to the MIMO detector. In this paper, we derive a basic framework to compare different soft-input soft-output MIMO detectors in open- and closed-loop systems. Within this framework, we analyze a depth-first sphere detector and a breadth-first fixed effort detector for different application scenarios and their effects on area and energy efficiency on the whole system. We present all system components under open- and closed-loop system aspects and determine the overall implementation cost for changing an open-loop system in a closed-loop system.

PENGARUH PENGGUNAAN CODEC STANDART ITU G.729 TERHADAP SISTEM KOMUNIKASI VOIP

2021 ◽  
Vol 17 (1) ◽  
pp. 11-22
Author(s):  
Wahyu Adi Prijono
Keyword(s):  
Communication System ◽  
Communication Systems ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Data Communications ◽  
Voice Communication ◽  
Internet Telephony ◽  
Voice Signal ◽  
Ip Network ◽  
Packet Switch

Voice over Internet Protocol (VoIP) is a technology that is capable of passing voice traffic, in the form of packets through the network Internet Protocol (IP). IP network itself is a data communications network based packet-switch. The voice signal before experiencing bundled voice coding or format conversion of sound into digital form that can be passed over an IP network. Telephony, Internet telephony, or termed VoIP (Voice Over Internet Protocol.This communication system use VoIP (Voice over Internet Protocol), ie voice calls over data services (internet). This communication was developed using Android-based devices. Based on characteristics, android devices are open source, so users do not need to have a license to be able to have android-based devices. In addition, the android device that must be connected to a SIP (Session Iniation Protocol) is a data service that can be done with a paid subscription of the user of the operator using a conventional pulse. Telecommunications designed will use a hybrid system, the merger between VoIP communications with data communications GSM network. With basic calculations where Coding standards G 729, is a standard that can be used for voice communication system through data networks with rate of 8 Kbps. The implementation of the G729 codec is effect on communication systems VOIP.

Interaction of Speech Coders and Atypical Speech II

2002 ◽  
Vol 45 (4) ◽  
pp. 689-699 ◽  
Author(s):  
Donald G. Jamieson ◽  
Vijay Parsa ◽  
Moneca C. Price ◽  
James Till
Keyword(s):  
Communication Systems ◽  
Speech Rate ◽  
Voice Quality ◽  
Voice Disorders ◽  
Normal Hearing ◽  
Speech Quality ◽  
Degraded Speech ◽  
Before And After ◽  
Subband Processing

We investigated how standard speech coders, currently used in modern communication systems, affect the quality of the speech of persons who have common speech and voice disorders. Three standardized speech coders (GSM 6.10 RPELTP, FS1016 CELP, and FS1015 LPC) and two speech coders based on subband processing were evaluated for their performance. Coder effects were assessed by measuring the quality of speech samples both before and after processing by the speech coders. Speech quality was rated by 10 listeners with normal hearing on 28 different scales representing pitch and loudness changes, speech rate, laryngeal and resonatory dysfunction, and coder-induced distortions. Results showed that (a) nine scale items were consistently and reliably rated by the listeners; (b) all coders degraded speech quality on these nine scales, with the GSM and CELP coders providing the better quality speech; and (c) interactions between coders and individual voices did occur on several voice quality scales.

scholarly journals Optimising outcome assessment of voice interventions, II: sensitivity to change of self-reported and observer-rated measures

2007 ◽  
Vol 122 (1) ◽  
pp. 46-51 ◽  
Author(s):  
I N Steen ◽  
K MacKenzie ◽  
P N Carding ◽  
A Webb ◽  
I J Deary ◽  
...  
Keyword(s):  
General Health ◽  
Voice Quality ◽  
Voice Handicap Index ◽  
Effect Sizes ◽  
Voice Therapy ◽  
Vocal Performance ◽  
Symptom Scale ◽  
Wide Range ◽  
The Voice ◽  
Responsiveness To Change

AbstractObjectives:A wide range of well validated instruments is now available to assess voice quality and voice-related quality of life, but comparative studies of the responsiveness to change of these measures are lacking. The aim of this study was to assess the responsiveness to change of a range of different measures, following voice therapy and surgery.Design:Longitudinal, cohort comparison study.Setting:Two UK voice clinics.Participants:One hundred and forty-four patients referred for treatment of benign voice disorders, 90 undergoing voice therapy and 54 undergoing laryngeal microsurgery.Main outcome measures:Three measures of self-reported voice quality (the vocal performance questionnaire, the voice handicap index and the voice symptom scale), plus the short form 36 (SF 36) general health status measure and the hospital anxiety and depression score. Perceptual, observer-rated analysis of voice quality was performed using the grade–roughness–breathiness–asthenia–strain scale. We compared the effect sizes (i.e. responsiveness to change) of the principal subscales of all measures before and after voice therapy or phonosurgery.Results:All three self-reported voice measures had large effect sizes following either voice therapy or surgery. Outcomes were similar in both treatment groups. The effect sizes for the observer-rated grade–roughness–breathiness–asthenia–strain scale scores were smaller, although still moderate. The roughness subscale in particular showed little change after therapy or surgery. Only small effects were observed in general health and mood measures.Conclusion:The results suggest that the use of a voice-specific questionnaire is essential for assessing the effectiveness of voice interventions. All three self-reported measures tested were capable of detecting change, and scores were highly correlated. On the basis of this evaluation of different measures' sensitivities to change, there is no strong evidence to favour either the vocal performance questionnaire, the voice handicap index or the voice symptom scale.

Conversion from a non-indwelling to a Provox®2 indwelling voice prosthesis for speech rehabilitation: comparison of voice quality and patient preference

2003 ◽  
Vol 117 (10) ◽  
pp. 815-820 ◽  
Author(s):  
A. C. Vlantis ◽  
R. T. Gregor ◽  
H. Elliot ◽  
M. Oudes
Keyword(s):  
Patient Satisfaction ◽  
Prospective Study ◽  
Patient Preference ◽  
Acoustic Analysis ◽  
Voice Quality ◽  
Voice Prosthesis ◽  
Technical Aspects ◽  
Speech Rehabilitation ◽  
Perceptual Difference ◽  
The Voice

This prospective study assessed the advantages and problems associated with converting a patient using an older generation non-indwelling voice prosthesis to a newer generation indwelling voice prosthesis, in this case the Provox®2. The voice characteristics of each patient were measured using the old and then the new voice prosthesis. Technical aspects of the insertion of the indwelling prosthesis were noted. Each patient completed a questionnaire after a period of use with the indwelling prosthesis.Changing the prosthesis was simple and uncomplicated in 15 of 17 patients. Acoustic analysis showed improved parameters with the indwelling prosthesis, but no perceptual difference between the two prostheses. The questionnaire revealed that most patients preferred the indwelling prosthesis.Replacing a non-indwelling with an indwelling prosthesis is technically simple, leading to improvement in voice quality and patient satisfaction. It may be reasonable to offer this choice to patients currently using an older generation non-indwelling voice prosthesis.

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