Time Assignment Speech Interpolation Research Articles

On some performance characteristics for a TASI system

The performance of a Time Assignment Speech Interpolation (TASI) system is usually characterized by the fraction of the freeze-out talkspurts of talkers. In the present investigation, we analyze the process of information transmission in a TASI system in details and introduce a new performance characteristic of such systems. Explicit analytical expressions are obtained under the assumption that input subscribers are continuously occupied, talkspurts and silence intervals are exponentially distributed.

Digital Speech Interpolation Techniques

The problem of optimum utilization of a link of finite capacity has been of interest for a long time. The major effort in this direction, resulted in the Time Assignment Speech Interpolation (TASI) technique albeit a relatively poor method. The drawback was not so much the basic idea, but the difficulty of implementation. The advances in Digital Coding and Digital Signal Processing (DSP) techniques have helped overcome most of the hurdles, and opened up possibilities for other techniques that would not have been possible with the conventional analog methods. Today we have a host of such methods, referred to generically as Digital Speech Interpolation (DSI) systems. In this paper we review some of the better known methods including Campanella's DSI which was the forerunner of the DSI era and present results of some of the recent DSI techniques implemented by the authors. Increase in channel capacity by a factor of at least 2 times can be achieved with little or no loss in quality. With a marginal reduction...

Performance analysis of a packet-switched synchronous voice/data transmission system

The queuing behavior of voice and data traffic for a proposed voice/data multiplexing system is investigated. For synchronization of packetized voice in the packet-switched network, the master frame format of time-division multiplexing (TDM) is utilized. This scheme has discrete delay characteristics for packetized voice and provides a simple play-out method for voice signals. The performance of voice and data traffic is investigated by obtaining the cumulative distribution of delay of voice packets and the mean delay time of voice and data packets. The performances of the proposed system and the circuit-switched system are compared to time-assignment speech interpolation (TASI) in terms of the loss probability of voice traffic and the maximum allowable number of input voice channels for a given trunk capacity. The proposed system has been found to be more efficient than the circuit-switched system with TASI for integrating multiple voice channels in a single link. For a given trunk capacity, the system can accommodate about twice the number of input voice channels as the circuit-switched system.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Performance Trade-Offs in an integrated voice/data services TDM system

A fairly tight upper bound for the mean queue size of a voice/data integrated TDM is derived. The result is applied to a queueing analysis of the data traffic subject to a prescribed voice call blocking probability. The analysis also indicates that the integration of voice and data services using time assignment speech interpolation (TASI) significantly improves the system performance over one without TASI. The problem of how to select the speech activity detection threshold at the voice source is addressed. We show that the voice correlation effect is heavily dependent upon the speech activity detection threshold at the voice source. A low threshold gives a low voice queueing delay and, therefore, a smaller time distortion on the received voice process.

Speech–nonspeech detector-classifier

The classification of a signal on a telephone line can be based on its short term average energy level; nonspeech (data or signalling) as high level; speech as medium level, and noise as low level. A common use for speech detectors occurs with a time assignment speech interpolation (TASI) system. A speech detector extends a trunk activity signal (TAS) for indicating whether energy or noise is detected on a trunk. Responsive to the TAS signal, a processor processes a signal on the trunk. On occasion, it may be desirable to process different energy signals according to different methodologies. For example, during overload, a TASI system may truncate bits of a speech signal and not truncate bits of a nonspeech signal. For classifying energy signals as between speech signals and nonspeech signals, the instant classifier includes an arrangement for extracting at least one feature, or property, such as the short term power of the signal to be classified. Responsive to the extracted property, an estimator provides a coarse estimate as to whether the signal to be classified is a speech signal or a nonspeech signal. Responsive to the estimate, a six-state, sequency machine provides a fine, classifying signal for classifying the energy signal.

PLC-1: A TASI System for Small Trunk Groups

The PLC-1 Private Line Voice Concentrator is introduced. This equipment uses the well-known TASI (time assignment speech interpolation) technique to reduce the number of transmission facilities required to transmit a number of trunks by a factor of up to two. In addition to reducing transmission facility requirements, the PLC-1 provides a very high level of management and diagnostic tools to the telecommunications manager.

Operational Evaluation of a Voice Concentrator Over AUTOVON Interswitch Trunks

This paper describes results of test and evaluation of a commercial voice concentrator applied to AUTOVON interswitch trunks between Feldberg, Germany and Ft. Detrick, MD. By virtue of time assignment speech interpolation (TASI) techniques, the voice concentrator provided approximately a 2-to-1 compression of voice channels onto trunks, with a configuration of 17 channels onto nine trunks selected for this AUTOVON application. Tests consisted of: 1) signaling performance characterization through the voice concentrator, 2) voice channel characterization of the nine selected trunks before and after cutover of the voice concentrator, 3) performance characterization of data signals operated through voice concentrator channels, 4) traffic data collection and analysis, and 5) user subjective evaluation. Test results indicated acceptable performance for the intended application, which was limited primarily to voice signals. Since completion of the testing and subsequent analysis, the Defense Communications Agency (DCA) has accepted the voice concentrator system for operational use with AUTOVON IST's in the Defense Communications System (DCS).

Simulation Studies for Data Multiplexing in Speech with Stdm or Tasi

A simulation model of a finite buffer queueing system with Erlang arrivals and multiple random servers is presented. The number of active servers in a service interval vary through a birth and death process derived from the speech samples in synchronous time division multiplexing (STDM) or time assignment speech interpolation (TASI) system. The results of this study show that the TASI system requires a larger buffer than the STDM system for a specified overflow probability.

The Design of an ADPCM/TASI System for PCM Speech Compression

This paper describes the design of a digital speech interpolation (DSI) system called ADPCM/TASI for adaptive differential PCM with time assignment speech interpolation. This system is designed to compress the output of two T1 24-channel PCM carrier terminals into a 1.544 Mbit/s signal that can be transmitted over a single T1 carrier line. The design is based on a bit slice microprocessor structure. Alternative designs are also described.

Performance of an ADPCM/TASI system for PCM speech compression

This paper contains a summary of performance measurements for a digital speech interpolation (DSI) system called adaptive differential PCM with time assignment speech interpolation (ADPCM/ TASI). This system is used to compress the output of two or more T1 PCM terminals into a bit rate normally requited by one such terminal. The performance measurements presented are as follows. 1) Signal-to-noise power ratios (SNR) for a 1004 Hz tone signal measured by a transmission impairment measuring test set (TIMS). 2) Idle channel noise performance measured by a TIMS. 3) SNR for speech measured by actual computation. These measurements were made for a variety of traffic loading conditions, and a number of different input power levels. The ADPCM/TASI system was constructed by using an AMD 2900 bit slice microprocessor architecture, but was designed to operate on only one speech channel. The presence of other active channels was simulated programatically. The performance of the system when voice band data signals occupy some of the T1 channels is discussed and is reflected in the measurements. An audio tape was made to demonstrate the subjective performance of the system with and without errors in the transmission medium. The objective performance of the system is compared to the D3 channel bank specifications.

Time assignment speech interpolation system—a microprocessor based design

A normal synchronous time multiplexed system has low capacity utilisation of a trunk channel, owing to the ON-OFF nature of speech in human conversation. A time assignment speech interpolation (TASI) system nearly doubles this efficiency, at relatively small increase in hardware cost. In TASI, during the silent period of speech, the channel is allotted to some other ‘active’ subscriber on a first-come-first-serve basis. A microprocessor based system offers a very cost effective solution from hardware count. Intel 8085 A has been selected for the purpose. The microprocessor functions in a distributed processing mode together with the main CPU controlling the stored program exchange. The system uses digital dynamic speech detectors for detecting transitions in speech over a channel, as they show considerably superior performance over amplitude detectors. An assembly language program for the system has been developed.

A finite waiting room queueing model with multiple servers having Markovian interruptions and its application to computer communications

A queueing model with finite waiting room, Poisson arrivals, multiple synchronous outputs and the number of active servers varying through the birth and death process is studied. Variation of active servers is obtained by providing a switch in each server. The switches are controlled through the birth and death process having finite population. The relationships between overflow probabilities, buffer size and expected queueing delay due to buffering are obtained. An efficient algorithm for computation of steady state probabilities is developed. Two digital voice-data systems; Data multiplexing in speech with synchronous Time Division Multiplexing (STDM) and Time Assignment Speech Interpolation (TASI) systems are considered for the application of the queueing model studied. The results of this study are portrayed on graphs and may be used as guidelines for designing a buffer in digital systems. The two voice-data systems using the developed model are simulated on the EC-1030 computer to check the validity of the analytical results. Although this problem arose in the study of data interpolation in STDM and TASI systems, the queueing model developed is quite general and may be useful for other industrial applications.

Three-state Markov model of speech on telephone lines and optimal utilization of communication systems by TASI technique

A three-state Markov model of speech on the telephone lines is developed. The model considers the alternate occurrence of the telephone calls and the intercall gaps on the telephone lines. During a phone call there are several talkspurts (speech without break) and pauses (silence without break) in the user's speech. The three types of events, intercall gaps (large gaps) talkspurts and pauses occurring on the telephone lines are assumed to have negative exponential density functions with different transition rate parameters. The steady state probability distribution, average and variance of the number of busy channels and the system utilization are evaluated as a function of call loss fraction ( P 0 f ). The Synchronous Time Division Multiplexing (STDM) system with number of channels in the group equal to 6 and 24 is considered. The model is also applied to Time Assignment Speech Interpolation (TASI) system and the relationships among the number of user terminals, speech Freeze Out Fraction (FOF) and system utilization are obtained for various values of P 0 f . The STDM and TASI systems applied to the group of 6 channels are simulated on the EC-1030 computer to check the validity of the analytical results. The results of this study are portrayed on graphs and may be used as guide lines in the design of TASI systems.

Variable Length Coding for Increasing Traffic Capacity in PCM Transmission Systems

The probability density function of PCM encoded speech signals has been measured, and it has been found that, as a result of companding, the distribution is rather flat except in the neighborhood of the zero level. On the basis of the measured probability density function (PDF) a variable length prefix code was developed that allows an increase in the traffic capacity of PCM transmission systems. The benefit obtained is mainly due to the high probability of the zero level. According to the proposed coding method, inactive channels are maintained by sending only one code bit per frame; they thus can be activated without delay and without signaling. This is an advantage with respect to the time assignment speech interpolation (TASI) system [1].In the paper the capability of this method has been investigated.

A Speech Predictive Encoding Communication System for Multichannel Telephony

This paper describes a speech predictive encoding communications (SPEC) system that is designed to accommodate the traffic of <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N</tex> PCM telephone trunks in the capacity of <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N/2</tex> PCM telephone channels. The system exploits the statistics of multichannel voice communication in a manner significantly different from the technique employed in the well-known time-assignment speech interpolation (TASI) system. Specifically, the SPEC system operates at the transmitter by removing redundant speech samples during talk bursts as well as during silence intervals. The receiver remembers the most recent sample transmitted through each channel. As long as a new sample is not transmitted, the receiver simply repeats the sample value stored in its memory. Each sample from each incoming trunk is examined every 125 μs to determine which samples must be transmitted and which are redundant. Because the SPEC system operates at this high speed and predicts redundant or missing samples at the receiver, it does not produce the speech clipping effects typical of TASI-like systems. Any voice-quality degradation due to increased call load and/or speech activity is manifested as an increase in noise power. The subjective result is a graceful degradation that is virtually imperceptible to the listener.

TTT System (50 Mbits/s PCM-TDMA System With Time Preassignment and TASI) and Its Satellite Test Results

An experimental 50 Mbits/s pulse-code modulation-time-division multiple access (PCM-TDMA) system with time preassignment and time-assignment speech interpolation (TASI) features (called the TTT system) was built for use in a satellite test. This system was designed for application in the INTELSAT IV network and has the channel capacity of more than 700 telephone channels. This capacity can be increased up to twice when pulse-code modulation-time-assignment speech interpolation (PCM-TASI) is adopted. The field test was conducted via INTELSAT III in 1970 and satisfactory results were obtained, which encourages the introduction of the TDMA system for commercial use.

Variable Length Words-Pulse Code Modulation

After a very brief resume of conventional pulse code modulation (PCM) features, this paper describes the operating principles of a variable length words - pulse code modulation system (VLWPCM). This new concept is a combination of TASI and conventional PCM. Time assignment speech interpolation or TASI is used on the very long routes, such as transoceanic cables, to improve the yield of the circuits at the cost of a very sophisticated terminal. The combination with PCM allows us to simplify the terminals in such a way that it can yield a lower cost. A coder and a decoder are described.

Time-assignment speech interpolation in time-compression-multiplex transmission

In time-compression multiplex (t.c.m.) transmission, the signals of the channels are scanned and the samples are stored before being transmitted at high speed. In this way, the signals of a large number of channels can be sent over a single transmission path. The number of telephone channels that can be transmitted can be at least doubled by the process called time-assignment speech interpolation (t.a.s.i.). This exploits the fact that, on average, each channel in a multiplex telephone system is transmitting speech for less than half the time. The addition of t.a.s.i. facilities to a t.c.m. system is relatively simple because the main t.a.s.i. requirement (switching of all speech signals) already exists as an inherent part of any t.c.m. system. Furthermore, the extra information which has to be transmitted (the channel identification information) can be incorporated in the standard t.c.m. signal with very little change in the bandwidth requirements. The simplest t.a.s.i. systems favour the lower-numbered channels, so that, in the event of more channels being active than there are transmission slots available, the same higher-numbered channels are always `frozen out?. In order to overcome this difficulty, a queue-control system is necessary. It can be made to balance the overload conditions such that the queuing channels always accept the back of the queue in time rotation. A queue controller for a t.c.m.-t.a.s.i. system consists of a logical array of conventional binary stores, gates and shift registers.

Discussion on “Time-compression-multiplex transmission”, “Gated capacitor store for t.c.m. transmission” and “Time-assignment speech interpolation in time-compression-multiplex transmission”

Discussion on “Time-compression-multiplex transmission”, “Gated capacitor store for t.c.m. transmission” and “Time-assignment speech interpolation in time-compression-multiplex transmission”

Over-All Characteristics of a TASI System

TASI (Time Assignment Speech Interpolation) has been in service on transatlantic submarine cable channels since mid-1960. Measurement of service quality on one TASI system (White Plains-London) indicates that system performance equals or exceeds the original engineering objectives in all but a few cases. Field modifications now being made should bring these exceptions into closer agreement with objectives. A companion paper <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">4</sup> discusses in detail the design considerations for TASI speech detectors and describes subjective tests made to determine the maximum permissible loading of TASI circuits without impairment of service.