Telephone Set Research Articles

A bi-phase enabled serial acquisition system for remote processing of digitized ECG

This paper presents an approach for acquiring digitized ECG samples in a personal computer using an offline communication technique. For this, two standalone embedded modules are placed at the two ends of the communication link. The transmit end module collects ECG samples from the source and stores them in a RAM. Finally it converts each data byte into a bi-phase encoded bit stream for transmission using a standard telephone set through post office telephone line. The receive-end module, coupled with the telephone receiver decodes the ECG data, and then delivers them to a desktop computer through the serial port. An application software in the computer is used to store these samples for visual inspection. The detailed architecture and test results are discussed with synthetic ECG data available from PTB diagnostic ECG database (ptb-db) under Physionet.

Formal Specification of Control Software Systems using Behavioral Views

Large-scale control software systems are generally complex to describe, construct, manage, understand, and maintain. This paper approaches to reducing this complexity separate software structural and behavioral descriptions. Much research about control software systems continues on software structures and their patterns, characterizations, and classifications. Currently, research on the behavioral aspect of control software systems includes using formal notations for specifying software behaviors. Large formal specifications, however, can be difficult to create and to understand; more research is needed into methods for assisting software requirements engineers in reducing these difficulties. This paper uses the idea of a software behavioral view. We believe that a fully developed methodology based on views would significantly reduce the complexity of creating and understanding software requirements. This paper deals with the use of view formalism, a state chart based formalism, to specify of control logic for a telephone set system. Keywords: Formal method; requirement specification; control system software; behavioral views and statecharts;

Design of Frequency Discriminator for Ringer Using 0.18 μm CMOS Technology

Ring frequency discriminator is an electronic system that can check the frequency of any incoming signal whether it is in the frequency band or not. It is also one of the features for CMOS Speech and Ringer integrated circuit (IC) that can perform all the necessary speech and line interface functions for telephone sets. The signal is considered as a valid signal if it has frequencies between 20 Hz and 60 Hz (20 Hz < ring frequencies < 60 Hz). For future work, it is also possible to design a ring melody generator. Once the valid ring signal is detected and is present for about 75 ms (depends on the designer) continuously, the melody generator will be activated, the ring signal will be monitored continuously and the melody generator will be immediately turn on or off according to the momentary presence of a valid or invalid ring signal respectively until next power on reset or off-hook. This work is implemented using Silterra 0.18 μm CMOS technology.

A Rural Next Generation Network (R-NGN) and Its Testbed

Rural Next Generation Networks (R-NGN) technology allows Internet protocol (IP) based systems to be used in rural areas. This paper reports a testbed of R-NGN that uses low cost Ethernet radio links, combined with media gateways and a softswitch. The network consists of point-to-point IP Ethernet 2.4 GHz wireless link, IP switches and gateways in each community, standard copper wires and telephone sets for users. It uses low power consumption, and suitable for low density users. This combination allows low cost systems as well as multiservices (voice, data, and multimedia) for rural communications. An infrastructure has been deployed in two communities in Cipicung Girang, a village 10 km outside Bandung city, Indonesia. Two towers link the communities with a network of Institut Teknologi Bandung (ITB) campus. In addition, local wirelines connect community houses to the network. Currently there are four houses connected to each community node (for a total of eight house), upon which we can perform various tests and measurements.

Hand held telephone set and audio processing method

To automatically set an optimal signal level of an ambient sound according to earphone type even in the case where either of both-ear mount type and single-ear mount type earphones is mounted to a telephone set main body unit, and to recognize the ambient sound even in the case where a stereo headphone almost covering ears is used. A hand held telephone set used by connecting a both-ear mount type or single-ear mount type earphone or thereto is provided with: a telephone set main body unit; detection means for detecting which of these earphones and is mounted to the telephone set main body unit; an amplifier for amplifying an ambient sound to be delivered to these earphones and; and a control device for controlling a signal level of the ambient sound output from this amplifier according to an output of the detection means. When either of these earphones and is mounted, the control device controls the signal level of the ambient sound output from the amplifier to be fed back according to an output of the detection means.

Toward Robust Speech Recognition and Understanding

The principal cause of speech recognition errors is a mismatch between trained acoustic/language models and input speech due to the limited amount of training data in comparison with the vast variation of speech. It is crucial to establish methods that are robust against voice variation due to individuality, the physical and psychological condition of the speaker, telephone sets, microphones, network characteristics, additive background noise, speaking styles, and other aspects. This paper overviews robust architecture and modeling techniques for speech recognition and understanding. The topics include acoustic and language modeling for spontaneous speech recognition, unsupervised adaptation of acoustic and language models, robust architecture for spoken dialogue systems, multi-modal speech recognition, and speech summarization. This paper also discusses the most important research problems to be solved in order to achieve ultimate robust speech recognition and understanding systems.

Concurrent data transmission through analog speech channel using data hiding

In this letter, we propose to perform concurrent data transmissions through analog speech channels using a data hiding technique. In particular, concurrent data are modulated into a noise-like signal and imperceptibly embedded into voices. Conventional embedding methods usually treat speech as a source of interferences to the simultaneously transmitted data. The proposed scheme exploits the knowledge of the speech signal during embedding. Therefore, such an interference can be effectively removed, and a higher data rate can be achieved. Besides, the proposed scheme is compatible with conventional phone terminals. A conventional telephone set is still able to access the basic voice service, while the added decoding mechanism can extract the embedded data and provide various concurrent services. The performance of the proposed method is analyzed. Experimental results demonstrate that an acceptable data rate and bit-error rate can be achieved.

Long-Term Health-Related Quality of Life, Growth, and Spiritual Well-Being After Hematopoietic Stem-Cell Transplantation

To examine health-related quality of life (HRQOL) and growth, and spiritual well-being in adult survivors of hematopoietic stem-cell transplantation (HSCT) for a malignant disease. HSCT survivors (n = 662) were recruited through the International Bone Marrow Transplant Registry/Autologous Blood and Marrow Transplant Registry and were drawn from 40 transplantation centers. HSCT survivors completed a telephone interview and a set of questionnaires a mean of 7.0 years post-HSCT (range, 1.8 to 22.6 years). Study measures included a variety of standardized measures of HRQOL and growth and spiritual well-being. An age- and sex-matched healthy comparison (HC) group (n = 158) was recruited using a peer nomination method. The HC group completed a parallel telephone interview and set of questionnaires. Multivariate analysis of variance analyses found the HSCT survivor group reported poorer status relative to the HC group for all HRQOL outcome clusters including physical health, physical functioning, social functioning, psychological adjustment, and dyadic adjustment. In contrast, the HSCT survivor group reported more psychological and interpersonal growth. Mean effect size for the 24 outcome indices examined was 0.36 standard deviations, an effect size often considered clinically meaningful or important. The largest group differences were found for measures of general health, physical function and well-being, depression, cognitive function, and fatigue. The experience of HSCT for a malignant disease has a wide-ranging, longstanding, and profound impact on adult recipients. Relative to healthy controls, HSCT survivors reported poorer physical, psychological, and social functioning but, conversely, more psychological and interpersonal growth, differences that appeared to persist many years after HSCT.

Audio signal amplifier circuit and portable telephone set and portable electronic device using the same audio signal amplifier circuit

An audio signal amplifier circuit includes an external output terminal of an IC to which an output line of a power amplifier circuit is connected, a first resistor connected between a certain terminal of the IC other than the external output terminal and a feedback input of a differential amplifier circuit, a first capacitor connected between the external output terminal and a loud speaker, a second capacitor between the certain terminal of the IC and a wiring line between the external output terminal and the loud speaker, a filter circuit provided on a signal input side of the differential amplifier circuit and including a second resistor and a third capacitor for attenuating signal components having frequencies in a middle and high frequency ranges and voltage follower means provided between an input stage and an output stage of the audio signal amplifier circuit, wherein the first capacitor is a small capacitor having a capacitance value in the order of 30 μF or smaller and an attenuation characteristics of the first capacitor and an impedance of the loud speaker in a low frequency range is compensated for by a relative amount of increase of amplification gain in the low frequency range determined by a time constant of the second capacitor and the first resistor and a time constant of the third capacitor and the second resistor.

Electromagnetic screening

A Bulgarian company has invented a new multi-layer metal coating deposited by chemical and electrochemical methods onto the surface acrylonitrile butadiene styrene-copolymer parts of the electronic device (e.g. telephone set). The individual layers can be either from a single metal (e.g. copper, nickel, iron), or alloys and composite materials as a polymer matrix containing highly dispersed fine-grained metal powder. The electromagnetic screening of electronic devices against unauthorised access is in conformity with US military standard-MIL-STD-461E/1999, applicable mainly in the telecoms and military industries. It is also used to protect equipment against extrinsic electromagnetic interference. The company is interested in licensing, joint venture and manufacturing agreement. This is a short news story only. Visit www.three-fives.com for the latest advanced semiconductor industry news.

Digital business telephones-project laser

In enhancing its 8400-series digital telephones used with the DEFINITY® private branch exchange (PBX), the Audio Display Terminals team decided to address simultaneously the challenges of reducing product design time and costs, and increasing manufacturing throughput and product reliability. In response to market management, the team — consisting of market management, research and development (R&D), and manufacturing in Lucent Technologies' Business Communications Systems (BCS) Business Unit — also needed to ensure that DEFINITY system customers could easily interchange the redesigned products with the earlier telephone sets, stipulating that the look and operation of the new products remain the same as before. The result was an initiative known as Project Laser, which implemented specific product improvements based on customer input, competitors' features, and three fundamental technology strategies. The design of the product line was optimized to support the new manufacturing process in the company's Shreveport, Louisiana, factory, and component integration was refined to provide future application flexibility. This paper discusses the Project Laser initiative, a breakthrough undertaking for BCS.

Telephone handset interface with feedback control

A telephone handset interface is disclosed for reducing or eliminating feedback induced singing in a telephone handset/hybrid loop. The interface provides for user selectable volume control settings, each setting representing an amount of gain or loss to be applied to the transmit and receive signals within the telephone set. The gain settings are selected to limit overall loop gain so as to reduce or eliminate any tendency of the handset to sing at high receive gain levels. Loss inserted into the transmit circuit is removed when user speech is detected in the transmit circuit.

State detection for hands-free telephone sets by means of fuzzy LVQ and SOM

The step gain of adaptive algorithms for acoustic echo cancellation should vary with time in dependence of the far-end signal, the local signal, and the adaptation quality. It can be described as state-dependent, which is also the case for the control parameters of other algorithms in a hands-free telephone set, such as adaptive loss control or noise reduction. Most of the known state detection methods can only distinguish between subsets of the states, and are unreliable for certain state transitions, so that they should be combined for improved reliability of the state detection. In this contribution, state detection is performed by a fuzzy learning vector quantization/self-organizing map approach to combine several state detectors. The step gain is inferred from state information using fuzzy logic concepts. Finally, adaptation results are compared to the best conventional step-gain control method used in the complete structure.

CTW: an integrated computer and telephone-accessed WWW system

This work presents the development of an integrated computer and telephone-accessed WWW system (CTW) to provide a ubiquitous web access service. The CTW system serves as an intermediary between the telephone user and web sites. The CTW system retrieves and then converts the proposed Hyper Phone Markup Language (HPML) specified web pages into speech patterns. Telephone users can access information, including form-based transaction services, on HPML-specified web pages using a telephone set anywhere in the world by connecting to the CTW system. Another feature of CTW is that HPML- specified web pages stored in cyberspace can also be accessed via regular browsers, including the current Netscape Navigator and Microsoft Internet Explorer. The CTW system provides a ubiquitous web access service for users at any degree of web-literacy by integrating well-installed telephone networks and the Internet. Copyright © 2000 John Wiley & Sons, Ltd.

Variations in the feedback of hearing aids.

Variations in the loop response of hearing aids caused by jaw movements, variations in acoustics outside the ear, and variations of vent size have been identified. Behind The Ear (BTE) and In The Ear Canal (ITEC) hearing aids were considered. The largest variations among the variations of the acoustics outside the ear, except when the hearing aid was partly removed, were found with the ITEC when a telephone set was placed by the ear. The variations of the loop response caused by changes in vent size were compared with the variations of a theoretical model of the feedback path. The theoretical model was also used to compare the feedback of different designs of the vent that gives the same acoustic impedance at low frequencies. The calculated feedback was less with the short vents (12 mm) than the long vents (24 mm).

Using experimental design to choose an acoustical model for a hands-free telephone

In full duplex speakerphones sound quality is aversely affected by the coupling between the speaker and the microphone. In addition to signal processing, this problem needs to be addressed by a better physical design of the telephone set. This requires the development of a computational model of the telephone acoustics. Due to the complexity of the geometry, the coupling between various systems and the rather large frequency band (150 Hz–7 kHz in wideband telephony), several complementary modeling approaches such as BEM/FEM, SEA, analytical or empirical methods are required. A determination of the best modeling technique for each phenomenon in each frequency band is needed. Instead of a classical one-factor-at-a-time method, an experimental design approach was used. This method goes beyond the qualitative interpretation of the experimental results and gives a statistical significance to the observed effects of each factor and their interactions. Although this approach is widely used in engineering, no reference was found in the literature of its proposed use to help with acoustical model selection. The method and its application using telephone set prototypes will be presented. [The support of MITEL, Inc. and NSERC is gratefully acknowledged.] a)Presently at MITEL, Inc., Kanata, Ontario.

Noise dosimetry for telephone users

Due to recent F.C.C. regulations, all telephones in North America will be required to have at least 12 dB of amplification. While this is a definite asset to the hard of hearing community, there are fears that normal hearing workers who use the telephone most of the day could suffer noise-induced hearing loss. Theoretical calculations validate this hypothesis. Measurements of actual levels in a call center were taken. Finally, a method of providing noise dosimetry within a digital telephone set is presented. Simultaneous measurements show good correlation.

The electrical century: a snapshot of telephony at the turn of the century

The telephone as we know it was slightly less than a quarter of a century old in 1900. There were just over 1.3 million telephones in service that year in the United States, handling an average of about 7.7 million conversations per day. Almost all telephone connections worldwide were handled by live operators using cord-and-plug switchboards. The idea of supplying subscribers' telephones with electricity from the central office had just been introduced, eliminating the battery that had previously sat under every telephone set. Calls of more than 10 miles or so were still considered long distance, and Pupin had recently described the "loading coil", an inductance device that extended long distance calling many more miles. Still, an east-coast caller could only reach out as far as Omaha, NE, by 1900. Telephone companies were rapidly building trunk lines between cities at the turn of the century, creating the new long-distance service. It became the practice, at AT&T at least, to charge for these calls by the minute, thus they were called "toll" calls. Operators used written tickets to indicate the destination of the call and its duration. The United States, where telephony got its commercial start, did not always have the highest level of telephone service in the early years. At the turn of the century, the percentage of the population with telephone service was considerably higher in Canada than either the United States (with only 17.6 telephones per thousand population) or England.

Speech signal transmitting and receiving apparatus with noise sensitive volume control

A speech signal transmitting receiving apparatus, such as a portable telephone set, includes a speech signal transmitting encoding circuit, a noise domain detection unit, a noise level detection unit and a controller. The speech signal transmitting encoding circuit compresses input speech signals by digital signal processing at a high efficiency. The noise domain detection unit detects the noise domain using an analytic pattern produced by the speech signal transmitting encoding circuit. The noise level detection unit detects the noise level of the noise domain detected by the noise domain detection unit. The controller controls the received sound volume responsive to the noise level detected by the noise level detection unit.

Noise reducing method, noise reducing apparatus and telephone set

A noise reducing method and device for reducing the noise contained in an input speech signal collects the speech signal with a microphone 11 and converts the speech signal into a digital input signal x(n) with an A/D converter 12. A frame power calculating circuit 13 calculates a mean frame power rms for each frame of the digital input signal x(n). A suppression ratio calculating circuit 14 calculates different values of the noise suppression ratio depending on the magnitude of the mean frame power rms relative to pre-set threshold values. A level discrimination circuit 18 forms a changeover control signal depending on the noise level and transmits the changeover control signal to the suppression ratio calculating circuit 14 for switching control of the threshold value. The suppression ratio value from the suppression value calculating circuit 14 is transmitted via a smoothing circuit 15 to a noise-reducing circuit 16 and multiplied with the input signal x(n) for reducing the noise component of the speech signal. The effect of the noise-reducing operation is changed in response to the noise level and the intensity of the noise-reducing operation is moderated in portions having a low noise level to prevent deterioration in the sound quality.