Comparison of Wideband Earpiece Integrations in Mobile Phone

Perinteisesti puhelinverkoissa välitettävä puhe on ollut kapeakaistaista, kaistan ollessa 300 - 3400 Hz. Voidaan kuitenkin olettaa, että laajakaistaiset puhepalvelut tulevat saamaan markkinoilla enemmän jalansijaa tulevina vuosina. Tässä lopputyössä esitellään puheenkoodauksen perusteet laajakaistaisen adaptiivisen moninopeuspuhekoodekin (AMR-WB) kanssa. Laajakaistainen puhekoodekki laajentaa puhekaistan 50-7000 Hz käyttäen 16 kHz näytetaajuutta. Käytännössä laajempi kaista tarkoittaa parannuksia puheen ymmärrettävyyteen ja tekee siitä luonnollisemman ja mukavamman kuuloista. Tämän lopputyön päätavoite on vertailla kahden eri laajakaistaisen matkapuhelinkuulokkeen integrointia. Kysymys kuuluu, kuinka paljon käyttäjä hyötyy isommasta kuulokkeesta matkapuhelimessa? Kuulokkeiden suorituskyvyn selvittämiseksi niille tehtiin objektiivisia mittauksia vapaakentässä. Mittauksia tehtiin myös puhelimelle pää- ja torsosimulaattorissa (HATS) johdottamalla kuuloke suoraan vahvistimelle, sekä lisäksi puhelun ollessa aktiivisena GSM ja WCDMA verkoissa. Objektiiviset mittaukset osoittivat kahden eri integroinnin väliset erot kuulokkeiden taajuusvasteessa ja särössä erityisesti matalilla taajuuksilla. Lopuksi tehtiin kuuntelukoe tarkoituksena selvittää erottaako loppukäyttäjä pienemmän ja isomman kuulokkeen välistä eroa käyttäen kapeakaistaisia ja laajakaistaisia puhelinääninäytteitä. Kuuntelukokeen tuloksien pohjalta voidaan sanoa, että käyttäjä erottaa kahden eri integroinnin erot ja miespuhuja hyötyy naispuhujaa enemmän isommasta kuulokkeesta laajakaistaisella puhekoodekilla.The speech in telecommunication networks has been traditionally narrowband ranging from 300 Hz to 3400 Hz. It can be expected that wideband speech call services will increase their foothold in the markets during the coming years. In this thesis speech coding basics with adaptive multirate wideband (AMR-WB) are introduced. The wideband codec widens the speech band to new range from 50 Hz to 7000 Hz using 16 kHz sampling frequency. In practice the wider band means improvements to speech intelligibility and makes it more natural and comfortable to listen to. The main focus of this thesis work is to compare two different wideband earpiece integrations. The question is how much the end-user will benefit from using a larger earpiece in a mobile phone? To find out speaker performance, objective measurements in free field were done for the earpiece modules. Measurements were performed also for the phone on head and torso simulator (HATS) by wiring the earpieces directly to a power amplifier and with over the air on GSM and WCDMA networks. The results of objective measurements showed differences between the earpiece integrations especially on low frequencies in frequency response and distortion. Finally the subjective listening test is done for comparison to see if the end-user notices the difference between smaller and larger earpiece integrations using narrowband and wideband speech samples. Based on these subjective test results it can be said that the user can differentiate between two different integrations and that a male speaker benefits more from a larger earpiece than a female speaker

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Network Traffic Aware Smartphone Energy Savings

In today\u27s world of ubiquitous Smartphone use, extending the battery life has become an important issue. A significant contributor to battery drain is wireless networking. Common usage patterns expect Smartphones to maintain a constant Internet connection which exacerbates the problem.;Our research entitled A Network Traffic Approach to Smartphone Energy Savings focuses on extending Smartphone battery life by investigating how network traffic impacts power management of wireless devices. We explore 1) Real-time VoIP application energy savings by exploiting silence periods in conversation. WiFi is opportunistically placed into low power mode during Silence periods. 2.) The priority of Smartphone Application network traffic is used to modifiy WiFi radio power management using machine learning assisted prioritization. High priority network traffic is optimized for performance, consuming more energy while low priority network traffic is optimized for energy conservation. 3.) A hybrid multiple PHY, MAC layer approach to saving energy is also utilized. The Bluetooth assisted WiFi approach saves energy by combining high power, high throughput WiFi with low power, lower throughput Bluetooth. The switch between Bluetooth and WiFi is done opportunistically based upon the current data rate and health of the Bluetooth connection.;Our results show that application specific methods for wireless energy savings are very effective. We have demonstrated energy savings exceeding 50% in generic cases. With real-time VoIP applications we have shown upwards of 40% energy savings while maintaining good call quality. The hybrid multiple PHY approach saves more than 25% energy over existing solutions while attaining the capability of quickly adapting to changes in network traffic

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

A novel approach to emergency management of wireless telecommunication system

The survivability concerns the service continuity when the components of a system are damaged. This concept is especially useful in the emergency management of the system, as often emergencies involve accidents or incident disasters which more or less damage the system. The overall objective of this thesis study is to develop a quantitative management approach to the emergency management of a wireless cellular telecommunication system in light of its service continuity in emergency situations – namely the survivability of the system. A particular wireless cellular telecommunication system, WCDMA, is taken as an example to ground this research.The thesis proposes an ontology-based paradigm for service management such that the management system contains three models: (1) the work domain model, (2) the dynamic model, and (3) the reconfiguration model. A powerful work domain modeling tool called Function-Behavior-Structure (FBS) is employed for developing the work domain model of the WCDMA system. Petri-Net theory, as well as its formalization, is applied to develop the dynamic model of the WCDMA system. A concept in engineering design called the general and specific function concept is applied to develop a new approach to system reconfiguration for the high survivability of the system. These models are implemented along with a user-interface which can be used by emergency management personnel. A demonstration of the effectiveness of this study approach is included.There are a couple of contributions with this thesis study. First, the proposed approach can be added to contemporary telecommunication management systems. Second, the Petri Net model of the WCDMA system is more comprehensive than any dynamic model of the telecommunication systems in literature. Furthermore, this model can be extended to any other telecommunication system. Third, the proposed system reconfiguration approach, based on the general and specific function concept, offers a unique way for the survivability of any service provider system.In conclusion, the ontology-based paradigm for a service system management provides a total solution to service continuity as well as its emergency management. This paradigm makes the complex mathematical modeling of the system transparent to the manager or managerial personnel and provides a feasible scenario of the human-in-the-loop management

Convivo communicator : an interface-adaptive voice over IP system for poor quality networks

Thesis (S.M.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2002.Includes bibliographical references (leaves 100-102).This thesis presents Convivo, a VoIP system designed to provide reliable voice communication for poor quality networks, especially those found in rural areas of the developing world. Convivo introduces an original approach to maintain voice communication interaction in presence of poor network performance: an Interface-Adaptation mechanism that suggests adjusting the application user interface to conform to one of three voice communication modalities (full duplex, half duplex, and voice messaging). The thesis proposes that changes in communication modality are an option to sustain voice communication interaction despite poor network performance. The goals of the changes in communication modality are to reduce the impact of high latency and low bandwidth on voice communication interaction, to facilitate turn taking for a high latency connection, and to sustain voice communication for extremely low bandwidth or high error links. The system was tested via a user study in Bohechio, a small village in the Dominican Republic. The study found that Interface-Adaptation helped users to maintain voice communication interaction when network performance degrades. Transitions from full duplex to voice messaging were found particularly valuable. Initial results suggest that as users get more experience with the application they would like to manually control transitions based on feedback provided by the application and their own perceived voice quality.by Marco Antonio Escobedo Gonzalez Maiz.S.M

Speech quality prediction for voice over Internet protocol networks

Merged with duplicate record 10026.1/878 on 03.01.2017 by CS (TIS). Merged with duplicate record 10026.1/1657 on 15.03.2017 by CS (TIS)This is a digitised version of a thesis that was deposited in the University Library. If you are the author please contact PEARL Admin ([email protected]) to discuss options.IP networks are on a steep slope of innovation that will make them the long-term carrier of all types of traffic, including voice. However, such networks are not designed to support real-time voice communication because their variable characteristics (e.g. due to delay, delay variation and packet loss) lead to a deterioration in voice quality. A major challenge in such networks is how to measure or predict voice quality accurately and efficiently for QoS monitoring and/or control purposes to ensure that technical and commercial requirements are met. Voice quality can be measured using either subjective or objective methods. Subjective measurement (e.g. MOS) is the benchmark for objective methods, but it is slow, time consuming and expensive. Objective measurement can be intrusive or non-intrusive. Intrusive methods (e.g. ITU PESQ) are more accurate, but normally are unsuitable for monitoring live traffic because of the need for a reference data and to utilise the network. This makes non-intrusive methods(e.g. ITU E-model) more attractive for monitoring voice quality from IP network impairments. However, current non-intrusive methods rely on subjective tests to derive model parameters and as a result are limited and do not meet new and emerging applications. The main goal of the project is to develop novel and efficient models for non-intrusive speech quality prediction to overcome the disadvantages of current subjective-based methods and to demonstrate their usefulness in new and emerging VoIP applications. The main contributions of the thesis are fourfold: (1) a detailed understanding of the relationships between voice quality, IP network impairments (e.g. packet loss, jitter and delay) and relevant parameters associated with speech (e.g. codec type, gender and language) is provided. An understanding of the perceptual effects of these key parameters on voice quality is important as it provides a basis for the development of non-intrusive voice quality prediction models. A fundamental investigation of the impact of the parameters on perceived voice quality was carried out using the latest ITU algorithm for perceptual evaluation of speech quality, PESQ, and by exploiting the ITU E-model to obtain an objective measure of voice quality. (2) a new methodology to predict voice quality non-intrusively was developed. The method exploits the intrusive algorithm, PESQ, and a combined PESQ/E-model structure to provide a perceptually accurate prediction of both listening and conversational voice quality non-intrusively. This avoids time-consuming subjective tests and so removes one of the major obstacles in the development of models for voice quality prediction. The method is generic and as such has wide applicability in multimedia applications. Efficient regression-based models and robust artificial neural network-based learning models were developed for predicting voice quality non-intrusively for VoIP applications. (3) three applications of the new models were investigated: voice quality monitoring/prediction for real Internet VoIP traces, perceived quality driven playout buffer optimization and perceived quality driven QoS control. The neural network and regression models were both used to predict voice quality for real Internet VoIP traces based on international links. A new adaptive playout buffer and a perceptual optimization playout buffer algorithms are presented. A QoS control scheme that combines the strengths of rate-adaptive and priority marking control schemes to provide a superior QoS control in terms of measured perceived voice quality is also provided. (4) a new methodology for Internet-based subjective speech quality measurement which allows rapid assessment of voice quality for VoIP applications is proposed and assessed using both objective and traditional MOS test methods

The Design of a System Architecture for Mobile Multimedia Computers

This chapter discusses the system architecture of a portable computer, called Mobile Digital Companion, which provides support for handling multimedia applications energy efficiently. Because battery life is limited and battery weight is an important factor for the size and the weight of the Mobile Digital Companion, energy management plays a crucial role in the architecture. As the Companion must remain usable in a variety of environments, it has to be flexible and adaptable to various operating conditions. The Mobile Digital Companion has an unconventional architecture that saves energy by using system decomposition at different levels of the architecture and exploits locality of reference with dedicated, optimised modules. The approach is based on dedicated functionality and the extensive use of energy reduction techniques at all levels of system design. The system has an architecture with a general-purpose processor accompanied by a set of heterogeneous autonomous programmable modules, each providing an energy efficient implementation of dedicated tasks. A reconfigurable internal communication network switch exploits locality of reference and eliminates wasteful data copies

Scalable Speech Coding for IP Networks

The emergence of Voice over Internet Protocol (VoIP) has posed new challenges to the development of speech codecs. The key issue of transporting real-time voice packet over IP networks is the lack of guarantee for reasonable speech quality due to packet delay or loss. Most of the widely used narrowband codecs depend on the Code Excited Linear Prediction (CELP) coding technique. The CELP technique utilizes the long-term prediction across the frame boundaries and therefore causes error propagation in the case of packet loss and need to transmit redundant information in order to mitigate the problem. The internet Low Bit-rate Codec (iLBC) employs the frame-independent coding and therefore inherently possesses high robustness to packet loss. However, the original iLBC lacks in some of the key features of speech codecs for IP networks: Rate flexibility, Scalability, and Wideband support. This dissertation presents novel scalable narrowband and wideband speech codecs for IP networks using the frame independent coding scheme based on the iLBC. The rate flexibility is added to the iLBC by employing the discrete cosine transform (DCT) and iii the scalable algebraic vector quantization (AVQ) and by allocating different number of bits to the AVQ. The bit-rate scalability is obtained by adding the enhancement layer to the core layer of the multi-rate iLBC. The enhancement layer encodes the weighted iLBC coding error in the modified DCT (MDCT) domain. The proposed wideband codec employs the bandwidth extension technique to extend the capabilities of existing narrowband codecs to provide wideband coding functionality. The wavelet transform is also used to further enhance the performance of the proposed codec. The performance evaluation results show that the proposed codec provides high robustness to packet loss and achieves equivalent or higher speech quality than state-of-the-art codecs under the clean channel condition

Channel Fading Statistics For Real-Time Data Transmission In Emergency Call Systems And Unmanned Aerial Systems

The Third Generation Partnership Project (3GPP) selected an in-band modem to transmit emergency data over cellular voice channel for the European Union emergency call (eCall) system. However, the road test results presented by the Harmonized eCall European Pilot project showed that the success rate of data delivery was only 71%, indicating that there is significant potential to improve its performance. In this dissertation, a testbed is designed for the eCall system that satisfies the 3GPP TS 26.267/268/269 standards. A method is proposed to measure the power of the received signal that passes through the in-band channel. Experiments are performed with the in-vehicle system testbed in a laboratory or a car travelling in city, suburb, country- side, or freeway. Fading statistics of the received signal after power control are found and discussed, together with cumulative distribution function (CDF), level crossing rate (LCR), and average fade duration (AFD). It is found that with probability less than or equal to 0.1%, fading and attenuation can vary from -19 dB for the continuous wave (CW) signal at 500 Hz to -9.5 dB for the CW signal at 2000 Hz. This dissertation recommends moving the CW signals at 500 Hz and 800 Hz for detection and synchronization in the 3GPP standard to 1500 Hz and 2000 Hz, respectively. This will give 9.5 dB improvement in detection and synchronization. The fading results are used to calculate the bit error rate (BER) performance for the eCall in-band modem. Synchronization detection probability are obtained by transmitting the synchronization preamble through various adaptive multi-rate vocoders and an additive white Gaussian noise channel. The testbed and proposed method are also used to measure the power of signals received by an unmanned aerial systems (UAS) and by the receiver in the operation center, respectively. Field experiments are carried out by flying the UAS above different locations. Statistics, including CDF, LCR, and AFD, are calculated for the six test-sites. The results of the fading statistics, synchronization detection probability, and BER can be directly applied to design real-time communication systems, including detection, delay estimation, modulation and coding