Real-time Voice Research Articles

Joint playout and FEC control for multi-stream voice over IP networks

Packet loss and delay are the major network impairments for transporting real-time voice over internet protocol (IP) networks. In the proposed system, multiple descriptions of the speech are used to take advantage of packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a joint playout buffer and forward error control (FEC) adjustment scheme that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that the proposed multi-stream voice transmission scheme achieves significant reductions in delay- and packet-loss rates as well as improved speech quality.

소규모 그룹에서의 음성 통신을 위한 TDMA 기반의 릴레이 프로토콜

Since the wireless communications have a limited transmission, the devices just around a master node can exchange data. Though Bluetooth and Zigbee support ad hoc, they are not appropriate for real-time voice communications. In this paper, we present a TDMA-based relay protocol for several users to communicate simultaneously. The proposed protocol can relay data or voice to other nodes in real-time by the multi-hop transmission method using TDMA. And the proposed protocol improves the network performance by allocating different frequencies to the slaves depending on the routing path scheduled by the routing table. NS-2 simulation shows that the performance of the proposed protocol is good in terms of the transmission delay and pecket loss probability in the real-time voice transmission.

Mobile Biometrics: Combined Face and Voice Verification for a Mobile Platform

The Mobile Biometrics (MoBio) project combines real-time face and voice verification for better security of personal data stored on, or accessible from, a mobile platform.

Real-Time Telehealth for COPD Self-Management Using Skype™

The utility of real-time interactive voice and video telehealth for teaching pursed-lips breathing (PLB) in chronic obstructive pulmonary disease (COPD) is unknown. This was a pilot study to determine its feasibility and efficacy on the key variables of social support and dyspnea. A randomized control study design with repeated measures (baseline, 4 and 12 weeks) was used. All participants in the control and intervention groups received PLB instruction at baseline, but only the intervention group received one weekly PLB reinforcement session for 4 weeks via home computer and Skype™ software. Outcome measures were Medical Outcomes Study Social Support Survey and dyspnea assessment (visual analogue scales for intensity and distress, modified Borg after six-minute walk distance, and Shortness of Breath Questionnaire for activity-associated dyspnea). A total of 22 participants with COPD (mean FEV1% predicted = 56) were randomized; 16 (9 telehealth, 7 control) completed the protocol. Intent-to-treat analysis at week 4, but not week 12, demonstrated significantly improved total social support (P = 0.02) and emotional/informational subscale (P = 0.03) scores. Dyspnea intensity decreased (P = 0.08) for the intervention group with a minimal clinical important difference of 10.4 units. Analysis of only participants who completed the protocol demonstrated a significant decrease in dyspnea intensity (P = < 0.01) for the intervention group at both week 4 and 12. Real-time telehealth is a feasible, innovative approach for PLB instruction in the home with outcomes of improved social support and decreased dyspnea.

Effects of iterative block ciphers on quality of experience for Internet Protocol Security enabled voice over IP calls

Voice over IP (VoIP) is the technology used to transport real-time voice over a packet-switched network. This study analyses the effects of encrypted VoIP streams on perceived Quality of Experience (QoE) from a user's perspective. An in-depth analysis on how the transparent nature of encryption can influence the way users perceive the quality of a VoIP call have been investigated by using the E model. A series of experiments have been conducted using a representative sample of modern codecs currently employed for digitising voice, as well as three of the most commonly used iterative block ciphers for encryption (DES, 3DES, AES). It has been found that the Internet Protocol Security encryption of VoIP strongly relates to the payload sizes and choice of codecs and this relationship has different effects on the overall QoE as measured by the E model, in terms of the way that users perceive the quality of a VoIP call. The main result of this paper is that the default payload shipped with the codecs is not the optimal selection for an increased number of VoIP calls, when encryption is applied and a minimum level of QoE has to be maintained, per call.

An Assessment of Security in Voip Using Secret Sharing

A major change in telecommunication industry is Voice over Internet Protocol (VoIP). VoIP offers interactive communications. It differs from conventional circuit switched networks. It allows people to communicate with each other at very low rates. The transmission of Real time voice data is not as easy as ordinary text data. The real time voice trans- mission faces lot of difficulties. It suffers from packet loss, delay, security and quality. These factors will affect the com- munication, degrades the performance and quality of a VoIP. This paper addresses the security aspects of VoIP to improve the quality.

Linear Prediction Analysis and Quantization for the Conjugate-Structure Algebraic-Code-Excited Linear-Prediction Speech Compression Algorithm

Voice-over Internet Protocol (VoIP) telephony which is the transmission of real-time voice over an Internet Protocol (IP) data network, is increasingly becoming a variant telecommunication technology that one day may surpass the old analog and digital telephone systems. The Quality-of-Service (QoS) factor is an important parameter to be considered when measuring the performance of a VoIP system. Algorithmic delay (latency) may influence the QoS of a VoIP system. This paper presents the hardware implementation of the Linear Prediction (LP) Analysis and Quantization component of the Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP) speech compression algorithms. The LP Analysis and Quantization units were implemented in hardware using Xilinx 11.1 ISE, after which ITU-T test vectors were used to determine whether they were equivalent implementations of the LP Analysis and Quantization. The algorithmic delays of hardware-implementations were obtained via simulation using Modelsim XE 6.4b while the algorithmic delays of the software-implemented speech compression algorithms were obtained by software profiling using the GPROF profiler. After comparison the algorithmic delays of the hardware-implementations of the LP Analysis and Quantization was found to be shorter than the algorithmic delays of their software-implemented and hardware-implemented counterparts.

Enhancing VoIP speech quality using combined playout control and signal reconstruction

The quality of real-time Voice over Internet Protocol (VoIP) networks is affected by network impairments such as delays, jitters, and packet loss. To solve this issue, this paper proposes a new receiver-based enhancing method of VoIP speech quality. Our approach is based on the combined playout control and signal reconstruction technique that consists of a set of algorithms that conceal packet loss, reduce buffering delay, detect spike delay, and alleviate packet delay jitter. The proposed fully receiver-based enhancing algorithm is computationally efficient, delivers high-quality voice service, and is suitable for use in any practical mobile VoIP system.

SmartCAC: Call Admission Control scheme to guarantee QoS for Voice over IEEE 802.11 WLANs

Most of the wireless LANs deployed today are IEEE 802.11b/g compliant. Since IEEE 802.11 standard specifies a contention based MACDCF, it cannot support QoS requirements of real-time traffic like voice or video. Thus these WLANs are not suitable for supporting real-time voice traffic. In this paper we propose an admission control scheme “SmartCAC”, which without modifying the basic access mechanism of IEEE 802.11 tries to guarantee QoS for voice traffic. SmartCAC is a channel BW allocation scheme; in which most of the available BW is reserved for real-time voice traffic and remaining small amount is allocated to non real-time data traffic. Moreover, the BW allocation is not static; rather it can be dynamically changed depending on traffic conditions. If the voice traffic is lightly loaded then rather than wasting the greater BW share allocated to voice, it is made available for non real-time traffic sources momentarily. Our ns-2 simulations result show that the proposed SmartCAC scheme can support strict QoS requirements such as high throughput and low delay (<150ms) required for voice over IP. General Terms Wireless Networks

Enhanced Timing Recovery Using Active Jitter Estimation for Voice-Over IP Networks

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called “jitter,” is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

“LISTENING AND REMEMBERING”: NETWORKED OFF-LINE IMPROVISATION FOR FOUR COMMUTERS

This paper analyses the experience of the networked off-line improvisation ‘Listening and Remembering’, a performance for four commuters using voices and sounds from the Mexico City and Paris metros. It addresses the question: how can an act of collective remembering, inspired by listening to metro soundscapes, lead to the creation of networked voice-and sound-based narratives about the urban commuting experience? The networked experience is seen here from the structural perspective (telematic setting), the sonic underground context, the ethnographic process that led to the performance, the narratives that are created in the electro-acoustic setting, the shared acoustic environments that those creations suggest, and the technical features and participants’ responses that pre- vent or facilitate interaction. Emphasis is placed on the participants’ status as non-performers, and on their familiarity with the sonic environment, as a context that allows the participation of non-musicians in the making of music through telematically shared interfaces, using soundscape and real-time voice. Participants re-enact their routine experience through a dialogical relationship with the sounds, the other participants, themselves, and the experience of sharing: a collective memory.

Open-Loop Pitch Analysis for the Conjugate-Structure Algebraic-Code-Excited Linear -Prediction Speech Compression Algorithm

Voice-over Internet Protocol (VoIP) telephony which is the transmission of real-time voice over an internet protocol (IP) data network, is increasingly becoming a variant telecommunication technology that one day may surpass the old analog and digital telephone systems. The Quality-of-Service (QoS) factor is an important parameter to be considered when measuring the performance of a VoIP system. Algorithmic delay (latency) may influence the QoS of a VoIP system. This paper presents the hardware implementation of the Open-Loop Pitch Analysis component of the Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP) speech compression algorithms. The Open-Loop Pitch Analysis units were implemented in hardware using Xilinx 11.1 ISE, after which ITU-T test vectors were used to determine whether they were equivalent implementations of the Open-Loop Pitch Analysis. The algorithmic delays of hardware-implementations were obtained via simulation using Modelsim XE 6.4b while the algorithmic delays of the software-implemented speech compression algorithms were obtained by software profiling using the GPROF profiler. After comparison the algorithmic delays of the hardware-implementations of the Open-Loop Pitch Analysis was found to be shorter than the algorithmic delays of their software-implemented and hardware-implemented counterparts.

The “One-Person Choir”: A Multidisciplinary Approach to the Development of an Embodied Human–Computer Interface

The One-Person Choir is a human-computer interface for singers that facilitates gestural control over a digital signal processing (DSP) mod- ule for harmonizing the singing voice in real time (see Figure 1). Harmonization adds extra pitch-shifted voices that are tonally related to the input voice. The interface captures global movements of the upper limbs by means of an integrated network of inertial sensors attached to the upper body of a singer. From these data, gestural cues are extracted and compared with a preconfigured gestural model that has been trained with empirical data. When the gestures of the singer match the preconfigured model, it is possible to control the harmonization of the singing input voice captured by a microphone. Thus, the interface allows a singer to naturally enhance the expressive qualities of his or her voice with the assistance of expressive gestures connected to an electronic

Power-aware opportunistic downlink scheduling in IEEE 802.16 wireless networks

The IEEE 802.16 is a standard for fixed and mobile Broadband Wireless Access (BWA). In this paper, we deal with two key challenges of 802.16-based networks. First, terminals close to cell edge experience poor channel quality, due to severe path-loss and high interference from concurrent transmissions in nearby cells. To address this issue, we propose a framework based on a static partitioning of bandwidth into chunks with different transmission power levels. Terminals with impaired channel conditions can then benefit from being allocated a higher amount of transmission power than the others. Secondly, transmissions should be scheduled according to Quality of Service (QoS) requirements to keep users with real-time video or voice calls satisfied, while best-effort connections should fairly share the remaining capacity. To this aim, we propose a scheduling algorithm, called Power-aware Opportunistic Downlink Scheduling (PODS), that aims at meeting both the QoS and fairness requirements, while taking into account the different power levels of the bandwidth chunks. The performance of the proposed scheduler is assessed through detailed packet-level simulation in realistic scenarios and compared with well-known scheduling algorithms. Results confirm that PODS is able to exploit power boosting to provide real-time connections with the desired level of QoS, irrespectively of their MSs’ channel quality.

A Weighted Feature Voting Approach for Robust and Real-Time Voice Activity Detection

This paper concerns a robust real-time voice activity detection (VAD) approach which is easy to understand and implement. The proposed approach employs several short-term speech/nonspeech discriminating features in a voting paradigm to achieve a reliable performance in different environments. This paper mainly focuses on the performance improvement of a recently proposed approach which uses spectral peak valley difference (SPVD) as a feature for silence detection. The main issue of this paper is to apply a set of features with SPVD to improve the VAD robustness. The proposed approach uses a weighted voting scheme in order to take the discriminative power of the employed feature set into account. The experiments show that the proposed approach is more robust than the baseline approach from different points of view, including channel distortion and threshold selection. The proposed approach is also compared with some other VAD techniques for better confirmation of its achievements. Using the proposed weighted voting approach, the average VAD performance is increased to 89.29% for 5 different noise types and 8 SNR levels. The resulting performance is 13.79% higher than the approach based only on SPVD and even 2.25% higher than the not-weighted voting scheme.

Optimal arytenoid adduction based on quantitative real‐time voice analysis

The optimal degree of arytenoid rotation for arytenoid adduction (AA) can be determined using quantitative real-time voice analysis. Repeated measures with each larynx serving as its own control. Unilateral vocal fold paralysis (VFP) was modeled in five excised canine larynges. Medialization laryngoplasty (ML) was performed, followed by AA. The optimal degree of arytenoid rotation was determined using real-time measurements of vocal efficiency (V(E) ), percent jitter, and percent shimmer. After the optimal degree of rotation was determined, the arytenoid was hypo- and hyperrotated 10% ± 2% of the optimal angle to mimic hypoadducted and hyperadducted states. Aerodynamic, acoustic, and mucosal wave measurements were recorded. Mean optimal angle of arytenoid adduction was 151.4 ± 2.5°. V(E) differed significantly across experimental conditions (P = .003). Optimal AA produced the highest V(E) of any treatment, but this value did not reach that produced in the normal condition. Percent jitter (P < .001) and percent shimmer (P < .001) differed across groups and were lowest for optimal AA. Mucosal wave amplitude of the normal (P = .001) and paralyzed fold (P = .043) differed across treatments. Amplitude of both folds was highest for optimal AA. V(E) and perturbation parameters were sensitive to the degree of arytenoid rotation. Using real-time voice analysis may aid surgeons in determining the optimal degree of arytenoid rotation when performing AA. Testing this method in patients and determining if optimal vocal outcomes are associated with optimal respiratory and swallowing outcomes will be essential to establishing clinical viability.

Telemonitoring in Heart Failure Rehabilitation

Exercise training (ET) is now recommended as an important component of a comprehensive approach to patients with heart failure (HF). Despite the existence of proven benefits of ET, many HF patients remain physically inactive. Introducing telerehabilitation (TR) may eliminate most of the factors that result in the currently low number of patients undergoing outpatient-based rehabilitation programmes and thus increase the percentage of those who will undergo cardiac rehabilitation. Despite the fact that TR is highly applicable and effective, there are few papers dedicated to the study of TR in HF patients. Until recently, only a couple of home rehabilitation-monitoring models have been presented, from the simplest, i.e. heart rate monitoring and transtelephonic electrocardiographic monitoring, through to the more advanced tele-electocardiogram (tele-ECG) monitoring (via a remote device) and realtime electrocardiographic and voice transtelephonic monitoring. It seems the last two are the most useful and reliable. Based on published studies, TR in HF patients could be equally effective as and provide similar improvements in health-related quality of life to standard outpatient cardiac rehabilitation. In addition, adherence to cardiac rehabilitation seems to be better during TR. Due to disease-related limitations, TR seems to be a viable alternative for comprehensive cardiac rehabilitation in HF patients. Further studies are needed to confirm the utility of this type of rehabilitation in routine clinical practice, including its cost-effectiveness. Because of the diversity of technological systems, it is necessary to create a platform to ensure compatibility between the devices used in telemedicine.

Methods for Detection of Speech Impairment Using Mobile Devices

Speech impairment is an important symptom of Parkinsons disease (PD). This paper presents a detailed systematic literature review on speech impairment assessment through mobile devices. A two-tier review methodology is utilized. The first tier focuses on real-time problems in speech detection. In the second tier, acoustics features that respond to medication changes in Levodopa responsive PD patients are investigated for recognition of speech symptoms. The investigation of the patents reveals that speech disorder assessment can be made by a comparative analysis between pathological acoustic patterns and the normal acoustic patterns saved in a database. The review depicts that vowel and consonant formants are the most relevant acoustic parameters to reflect PD medication changes. Since consonants have high zero-crossing rate (ZCR) whereas vowels have low ZCR, enhancements in voice segmentation can be done by inducing ZCR. Our synthesis further suggests that wavelet transforms have potential for being useful in real-time voice analysis for detection and quantification of symptoms at home.

Power-efficient scheduling for voice services in high-speed packet access systems

The high-speed downlink packet access (HSDPA) system can effectively support real-time voice services if the HS-SCCH-less feature is adopted in the 3GPP Release 7 standard. In this paper, we point out that the feature only offers a limited statistical multiplexing capability and it also causes serious power consumption at the user equipment. This paper proposes an HSDPA scheduling operation that utilizes the entire code space and considers the packet generation pattern of voice services.

Perceptual-Based Playout Mechanisms for Multi-Stream Voice over IP Networks

Packet loss and delay are the major network impairments for transporting real-time voice over IP networks. In the proposed system, multiple descriptions of the speech are used to take advantage of the packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a multi-stream playout buffer algorithm, together with an adaptive parameter adjustment scheme, that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that, compared to FEC-protected single-path transmission, the proposed multi-stream transmission scheme achieves significant reductions in delay and packet loss rates as well as improved speech quality.