Channel-dependent scheduling (CDS) can increase the data throughput of a cellular system by exploiting multi-user diversity and frequency selectivity in the channel. In this paper, we investigate the impact of imperfect channel state information (CSI) on CDS. Specifically, we analyze the data throughput of an uplink single carrier FDMA (SC-FDMA) system with uncoded adaptive modulation and CDS when there is a CSI feedback delay. We consider distributed and localized subcarrier mapping schemes for resource allocation. We show that localized subcarrier mapping yields highest aggregate data throughput when we use CDS. However, we also show that localized mapping is very sensitive to the quality of CSI and the capacity gain quickly decreases when the channel changes rapidly. For high mobility users, distributed mapping with static round-robin scheduling is more suitable.

In this work, we consider a CDMA cell with multiple terminals transmitting video signals. We adapt the system parameters to minimize the sum of compression powers and transmitter powers of all users while guaranteeing the received video quality at each terminal. The adjustable parameters at user i include the transmitter power Pt,i, the video coding bit rate Rs,i, and video encoder parameters that control the complexity and hence power consumption of the video coder (referred simply as complexity betai). Instead of determining Pt,idirectly, we first determine the desired signal to interference-noise ratio (SINR) gammai. Based on the optimal gammaiand Rs,i, we then determine Pt,i. Our analysis shows that the product of Rs,iand gammaiis an important quantity. Given the complexity betai(i.e., given the compression power) and quality constraint, in order to reduce the transmission power, one should choose Rs,iand gammaito minimize their product. When only the total transmission power is concerned, the optimal operating points can be determined at individual users separately: each user should run the encoder to minimize the product of Rs,iand gammai. When the objective is to minimize the sum of compression and transmission powers of all users, the optimal solution can be found in two steps. The first step searches the optimal Rs,iand gammaithat minimize Rs,itimesgammaifor each video category and each possible betaiwhile satisfying the quality constraint at user i. The second step searches the optimal {betai}i=1,...,Nfor all users jointly, that minimizes the sum of transmission and compression powers of all users. The first step can be completed offline in advance, only the second step needs to be computed in real time based on channel conditions of the users. Our results indicate that for the same class of video users, the one who is closer to the base station compresses at a lower complexity. Simulation results show that significant power savings are obtained by our adaptive algorithms over nonadaptive approaches, where {Rs,i,betai,gammai} are fixed regardless the channel conditions

This paper addresses uplink power control for terminals that transmit multimedia signals in a CDMA cell. The aim of the power control is to minimize total power consumed by the terminals due to signal compression and transmission while the end-to-end distortion for each terminal is kept at a predetermined value. We propose a distributed iterative power control algorithm and prove convergence. The simulations for Gauss-Markov source with transform coder and H.263 encoded video signal show that the proposed algorithm achieves the jointly optimal power values for most of the channel conditions. It is also observed that the algorithm outperforms non-adaptive multimedia transmission in terms of power consumption

This paper investigates the effects of time-scheduling for real time services with respect to power efficiency and data throughput of coexisting best effort services. In contrast to the best effort services, backlog delay due to buffering is a critical factor for real time services. Considering the randomness of user locations and maximum transmit power limit, we evaluate the performance gains of time scheduling obtained at the cost of linearly increasing backlog delay. Numerical examples show that the gain of time scheduling is small for low bit rate real time services requiring low latency (i.e. voice). Otherwise in a highly loaded cell, time scheduling of real time services can reduce intracell interference. This leads to significant improvements in power efficiency and throughput performance for both real time users and co-existing best effort users.

We apply novel utility-based scheduling schemes to uplink single carrier frequency division multiple access (SC-FDMA) systems. Two utility functions are used for managing two dimensional resources (time and frequency): user data rate for maximizing system capacity and logarithmic user data rate for proportional fairness. To develop utility-based scheduling algorithms, we revise channel-dependent scheduling (CDS) schemes derived in our previous work (J. Lim et al.). The results show that proportional fair scheduling with logarithmic user data rate can improve the rate-sum capacity up to 100% for localized FDMA and 30% for interleaved FDMA, with the capacity gains equally shared among all users

Single carrier frequency division multiple access (SC-FDMA), which utilizes single carrier modulation and frequency domain equalization is a technique that has similar performance and essentially the same overall complexity as those of OFDM, in which high peak-to-average power ratio (PAPR) is a major drawback. An outstanding advantage of SC-FDMA is its lower PAPR due to its single carrier structure. In this paper, we analyze the PAPR of SC-FDMA signals with pulse shaping. We analytically derive the time domain SC-FDMA signals and numerically compare PAPR characteristics using the complementary cumulative distribution function (CCDF) of PAPR. The results show that SC-FDMA signals indeed have lower PAPR compared to those of OFDMA. Comparing the two forms of SC-FDMA, we find that localized FDMA (LFDMA) has higher PAPR than interleaved FDMA (IFDMA) but somewhat lower PAPR than OFDMA. Also noticeable is the fact that pulse shaping increases PAPR

We examine single carrier frequency division multiple access (SC-FDMA) with frequency domain equalization for uplink data transmission. We investigate channel-dependent scheduling schemes to achieve multi-user diversity and frequency selective diversity. There are two subcarrier mapping schemes in SC-FDMA: Localized FDMA (L-FDMA) and Interleaved FDMA (I- FDMA). L-FDMA benefits from frequency selective scheduling, but it incurs higher peak-to-average power ratio than I-FDMA. Throughout our work, we provide low complexity channel- dependent scheduling (CDS) methods for L-FDMA and I-FDMA. The results show that rate-sum capacity can increase up to 130% for L-FDMA and 40% for I-FDMA relative to static round robin scheduling.

We consider a CDMA cell with multiple terminals transmitting video signals. We minimize the sum of signal processing and transmitter power while the received quality at each terminal is guaranteed. The system parameters to be adjusted include video coding bit rate, video compression complexity and transmitter power. Instead of full search in the space of {bit rate, complexity, transmitter power} for all users, we design a two-step fast algorithm to reduce the computation burden in the base station. In our algorithm, the search in the base station is over the space of complexity only. Our results indicate that, for the same class of video users, the one who is closest to the base station compresses at least complexity. This is used to further reduce the computation required by our algorithm.

Power control is a fundamental component of CDMA networks because of the interference that users cause to one another. Consequently, too many users in the system may lead to an overload whereas too few would generate an inefficient use of resources. Previous work by the authors has highlighted some fundamental properties of a CDMA system pertaining to the required power distribution when a particular terminal has reached its power limit. These properties have formed the basis for an admission control scheme which leads to an efficient use of system resources. This paper expands on this scheme and shows that optimal throughput with a fixed number of users can be achieved for a range of received power values and that this range of values is affected by the geometry of the users' location relative to the base station. In addition, there is an optimal number of users and an optimal received power ratio between them that results in maximum throughput. But when the optimal conditions can not be achieved, it is known that equal received power has good performance, and based on this, we design an admission control algorithm.

In this work, we consider a CDMA cell with multiple terminals transmitting video signals. The concept of a utility function is used to maximize the number of received picture frames with adequate quality per Joule of energy. For a reconstructed signal at the video decoder, the quality is controlled by the encoded bit rate, compression complexity as well as received signal-to-interference-noise ratio (SINR). In this work, video quality is measured in peak signal-to-noise ratio (PSNR) rather than SINR. We find that for a given compression complexity, maximum utility is achieved when the product of bit rate and required SINR is minimized. This maximum usually occurs at maximum video coding complexity. We also investigate the capacity in terms of the number of users that can be supported simultaneously for this system, and how the total utility varies with the number of users in this system. This can be used as an admission policy by the central station.

Scalably encoded information results in files which can be truncated at an arbitrary point and decoded, as supported by the JPEG-2000 (image) and MPEG-4 (video) standards. This work introduces a tractable, yet flexible analytical model for resource management involving scalably encoded video. Each segment of video of a predetermined length yields a file that can be truncated and decoded independently of other segments. The problem is set up as a joint optimization of transmission power, and coding rate (where to truncate?). The analysis reveals that any one of these variables uniquely determines the other. The terminal should truncate the file at the point that maximizes quality per unit of power employed.

We investigate the power consumed by two mobile terminals transmitting compressed source signals to a base station in a CDMA cellular system. The aim is to minimize total power consumption at the terminals by simultaneously adjusting the complexity of source compression and the transmitter power. We find that in general the complexity of source compression should increase with increasing distance between a terminal and the base station. However, the exact configuration of compression and transmitter power depends on the interaction of the two terminals. The optimum operating points of the two terminals are contained in a pair of nonlinear equations. We use the example of a transform coder processing signals from a Gauss-Markov source to explore the advantages of an adaptive system over a system with fixed compression. The analysis can be applied to a variety of source coders and be extended to a system with more than two users per cell.

We analyze the optimum transmitter power levels and the optimum number of active terminals sending data to a CDMA base station. The objective is to maximize the aggregate throughput of the base station. We find that in the presence of additive noise, received power balancing is suboptimal mathematically. We consider N terminals transmitting at the same data rate, with the power of the most distant terminal (terminal N), fixed at its maximum value, and the power of the other N-1 terminals varying. We conclude that the aggregate throughput at the base station is maximized or minimized (depending on the spreading gain G) when the receiver powers for the N-1 terminals operate at an identical receiver power, which is larger than that of the power-limited terminal. This finding reduces the complexity of the analysis to a univariate optimization problem. A numerical analysis indicates the extent to which additive noise reduces the optimum number of active terminals and the maximum base station throughput.

Relevant to the uplink of a VSG-CDMA system, a technique part of 3G standards, this work investigates power and data rate allocations that maximize the network weighted throughput. Each terminal has one of 2 possible weights, which admit various practical interpretations. Earlier works of ours tell us that at least one terminal should operate at the highest available data rate, and that terminals not operating at this rate should operate at the same signal-to-interference ratio (SIR). We have also learned that when only two terminals of dissimilar weights operate at maximal data rate, the optimal SIR values for these terminals is obtained as an intersection point between an "X-shaped" graph arising from optimality conditions, and a "U-shaped" graph arising from the feasibility condition on power ratios. In the present work, we introduce a general procedure to seek an allocation that is a global optimizer. The analysis is based on classical optimization theory, and should accommodate a wide variety of physical layer configurations.

Relevant to the uplink of a VSG-CDMA system, a technique part of 3G standards, this work investigates power and data rate allocations that maximize the network weighted throughput. Each terminal has one of 2 possible weights, which admit various practical interpretations. Earlier works of ours tell us that at least one terminal should operate at the highest available data rate, and that terminals not operating at this rate should operate at the same signal-to-interference ratio (SIR). This value is determined by the physical layer through the function that gives, in terms of the received SIR, the probability that a data packet is received correctly. In this work, we provide a more specific description of the optimal allocations. The key to the solution is an intersection point between an "X-shaped" graph arising from optimally conditions, and a "U-shaped" graph arising from the feasibility condition on power ratios. Our analysis is based on classical optimization theory, and should accommodate a wide variety of physical layer configurations.

Relevant to the uplink of a VSG-CDMA system, a technique part of 3G standards, this work seeks power and data rate allocation for each of the N terminals, so that the network weighted throughput is maximized. The weights admit various interpretations, including levels of importance, "utility", and price. We have learned that at least one terminal should operate at the highest available data rate. Our analysis leads to allocations in which terminals not operating at the highest data rate operate at the same signal-to-interference ratio (SIR). This value is determined by the physical layer through the function that gives, in terms of the received SIR, the probability that a data packet is received correctly. Other factors held constant, lowering the highest available data rate increases the number of terminals which should operate at maximum data rate. This analysis conforms to classical optimization theory. Our model should accommodate a wide variety of physical layer.

In this work, we introduce an approach for minimizing the total power consumption of a mobile transmitter due to source compression, channel coding and transmission subject to a fixed end-to-end source distortion. We illustrate our approach both on an abstract class of sources and channels and on a realistic H.263 video transmission system through a wireless channel. Performance under different channel environments and implementation schemes are investigated. Our numerical analysis shows that optimized settings can reduce the total power consumption by a significant factor and prolong battery life considerably compared with fixed parameter settings.

A major challenge in the operation of wireless communications systems is the efficient use of radio resources. One important component of radio resource management is power control, which has been studied extensively in the context of voice communications. With the increasing demand for wireless data services, it is necessary to establish power control algorithms for information sources other than voice. We present a power control solution for wireless data in the analytical setting of a game theoretic framework. In this context, the quality of service (QoS) a wireless terminal receives is referred to as the utility and distributed power control is a noncooperative power control game where users maximize their utility. The outcome of the game results in a Nash (1951) equilibrium that is inefficient. We introduce pricing of transmit powers in order to obtain Pareto improvement of the noncooperative power control game, i.e., to obtain improvements in user utilities relative to the case with no pricing. Specifically, we consider a pricing function that is a linear function of the transmit power. The simplicity of the pricing function allows a distributed implementation where the price can be broadcast by the base station to all the terminals. We see that pricing is especially helpful in a heavily loaded system.

The number of wireless Ethernet products based on the IEEE 802.11b technology has been growing rapidly for the past few years. Bluetooth is an emerging technology with the potential to equal the popularity of wireless local area networks (WLAN). Even though the systems are designed for different purposes, it is very likely for them to operate in the vicinity of each other. Since both systems operate in the same frequency band, this will cause interference between them. A model is developed that captures the performance impact of Bluetooth interference on the collocated IEEE 802.11b, parameterized by the data rate and packet size of 802.11b, the number of Bluetooth piconets, the piconet utilization, and the distance between the 802.11b and the Bluetooth radios. The analysis comprises interference at both physical and medium access control layers of the 802.11b systems.

We consider distributed power control in a multicell wireless data system and study the effect of pricing transmit power. Drawing on the earlier work of Goodman and Mandayam (see IEEE Personal Commun. Mag., vol.7, p.48-54, 2000), we formulate the QoS of a data user via a utility function measured in bits per Joule. We consider distributed power control, modeled as a noncooperative game, where users maximize their utilities in a multicell system. Base station assignment based on received signal strength as well as received signal-to-interference ratio (SIR) are considered jointly with power control. Our results indicate that for both assignment schemes, such a procedure results in an inefficient operating point (Nash equilibrium) for the entire system. We introduce pricing of transmit power as a mechanism for influencing data user behavior and our results show that the distributed power control based on maximizing the net utility (utility minus the price) results in improving the Pareto efficiency of the resulting operating point. Variations of pricing based on global and local loading in cells are considered as a means of improving the efficiency of wireless data networks. Finally, we discuss the improvement in utilities through a centralized scheme where each base station (BS) calculates the best SIR to be targeted by the terminals it is assigned.

Recent work has considered power control for wireless data networks from the framework of microeconomics. The user preferences have been mapped to utility functions measured in useful bits transmitted per unit of battery energy. We consider utility maximizing distributed power control in the presence of pricing and study the existence and efficiency of equilibria achieved. Specifically, we investigate the Pareto efficiency of a pricing policy that punishes asocial behavior in terms of the transmit power used by each user and hence the interference caused by it. The Nash equilibria achieved under pricing are characterized by using supermodularity. We propose a simple implementation of the pricing scheme where the base station broadcasts a pricing parameter that users use in their distributed power control algorithm. With the appropriate choice of the pricing parameter, all users increase their utilities significantly as compared to the case where there is no pricing. We also discuss the implication of pricing with regard to achieving a socially optimum operating point.

We develop a framework for power control for wireless data services based on economic concepts of utility and pricing. Our approach is based on a model for the level of satisfaction (utility) a wireless data user derives from using the system. Using this model, we present a distributed power control scheme that maximizes the utility of each user. Formulating this algorithm as a non-cooperative game, we show conditions for the feasibility of such power control as well as the existence and uniqueness of the Nash equilibrium achieved by the non-cooperative game. However, the resulting equilibrium is shown to be Pareto inefficient. A detailed analysis of this equilibrium provides motivation for introducing pricing in the system. A distributed power control scheme is then proposed where users maximize the difference in their utility and price. It is shown by way of simulation that by introducing pricing in the power control algorithm, there is a Pareto improvement in terms of the utilities that the users obtain. Numerical results are presented for the special case of a code division multiple access (CDMA) system for PCS (personal communication services) band.

In Packet Reservation Multiple Access (PRMA) protocol, transmission errors in the header of the packets and acknowledgements have a negative effect on capacity and speech quality. This paper reports a simulation study of the effects of errors on PRMA protocol stability and on speech quality. We also propose several changes to the protocol and introduce simple transmission techniques to improve the performance of the system. Based on these results, we identify a nominal PRMA system that provides twice the capacity of the original protocol in the presence of transmission errors.>

We report the performance of the CCITT standard G.721, adaptive differential pulse code modulation, when channel impairment causes erasure of frames of information. The structure of the G.721 decoder is analyzed to explain the behavior observed during frame erasures and when correct transmission is restored. Various techniques have been tried to the decoder when frames are lost and when correct transmission is restored. Though the decoder, at times, takes long to gain synchronism with the encoder, the amplitude of the output signal approaches the error-free output much more rapidly. This fast recovery of amplitude helps in reducing the impact on the perceptual quality of speech. From objective results and informal listening tests, it was found that simple silence substitution performs as well as other frame substitution techniques. Resetting the encoder and decoder simultaneously at the end of erasure is seen to improve the performance. Since silence substitution is the easiest to implement, this technique seems the most appropriate one for application to wireless communications. This work is targeted towards understanding the implications to local exchange networks of wireless system alternatives that could provide access to those networks.>

The capacity of cellular mobile radio systems can be increased by controlling the transmitter powers so as to balance the carrier to interference ratios at the receivers. A distributed algorithm for implementing such power control was proposed earlier for satellite systems by Meyerhoff (1974). The authors tighten the mathematical arguments and apply this scheme to cellular systems. The mobile units adjust their transmitter powers at discrete time instants. Specifically at each time instant the mobile multiplies its current power by a factor equal to a constant over its current carrier to interference ratio (CIR). So power is increased or decreased based on the value of the current CIR. After a number of such power adjustments all the mobiles converge to a common CIR. They also present numerical results which show that the above scheme has the potential to converge faster than the scheme proposed by J. Zander (see IEEE Transactions on Vehicular Technology, vol.41, no.3, August 1992) for high CIR values.>

Future wireless information networks will employ dense grids of base stations to support a high user population. While small cells mitigate capacity problems, lengthen battery life, and improve signal quality, the frequent movement of users across cell boundaries imposes new burdens on network controllers. Previous work suggests that a cellular packet switch, based on distributed network control, can support high density personal communications. This paper proposes an allocation of control functions to elements of the cellular packet switch and specifies the flow of information within the switch. Message flows, defined for key procedures, form the foundation of a network layer protocol. Functionality is divided into the categories of call processing (set up and release), mobility management (location updates, handover, and path optimization), and radio resource management. Central office switches perform call processing. Mobility management and radio resource management are performed by interface units of the metropolitan area network at the heart of the cellular packet switch. This architecture delivers advanced services with minimal impact on central office switch software. It also relies on distributed, rather than centralized mobility management to provide prompt, accurate communications as users move about.>

The future of third-generation wireless networking is discussed. The vision of the third generation is a single set of standards that can meet a wide range of wireless access applications. Third-generation systems, in harmony with broadband integrated services digital networks, will use shared resources to convey many information types. A single network architecture will serve its users efficiently in many environments, including moving vehicles, indoor and outdoor public areas, residences, offices, and factories. A study of a switching architecture, referred to as a cellular packet switch, and a packet transmission technique, referred to as a packet reservation multiple access is discussed. By means of a design example, it is shown how these techniques can work together to meet some of the demands of third-generation systems.>

Simulation work is reported indicating that packet reservation multiple access (PRMA) allows a variety of information sources to share the same wireless access channel. Some of the sources, such as speech terminals, are classified as periodic and others, such as signaling, are classified as random. Packets from all sources contend for access to channel time slots. When a periodic information terminal succeeds in gaining access, it reserves subsequent time slots for uncontested transmission. Both computer simulations and a listening test reveal that PRMA achieves a promising combination of voice quality and bandwidth efficiency.>

We consider a set of uncoordinated users employing an ALOHA protocol to transmit packets to a central base station in a local radio environment, e.g., in an urban area or within a building. Because of the different distances between the various users and the base station, the signals received from different users have substantially different power levels. This near/far phenomenon gives rise to a capture effect at the receiver such that when several packets are transmitted simultaneously, the receiver has a good chance of accurately detecting the packet arriving with the highest energy. Moreover, capture can be made near perfect in this environment through the use of properly design spread-spectrum signaling technique that enables the receiver to differentiate among the arriving packets based on both power differences and differing arrival times. We investigate variations to the conventional ALOHA protocols that take advantage of perfect capture to reduce the delays and increase the throughputs of all users, including the furthest one. Three protocols are introduced and compared, which we call "persistent ALOHA with capture" (PAC), "gated ALOHA with capture" (GAC), and "gated ALOHA with capture/exhausted" (GAC/E). Two goals are considered in the comparison: an equatable delay profile among the users, and a small expected number of simultaneously transmitted packets. The latter is quite relevant for the practical implementation of a perfect-capture receiver. We conclude that the GAC/E protocol is most desirable from the twin points of view of delay profile and receiver implementation.

Packet communication systems cannot, in general, guarantee accurate and prompt delivery of every packet. The effect of network congestion and transmission impairments on data packets is extended delay; in voice communications these problems lead to lost packets. This paper describes techniques for replacing missing speech with waveform segments from correctly received packets in order to increase the maximum tolerable missing packet rate. After presenting a simple formula for predicting the probability of waveform substitution failure as a function of packet duration and packet loss rate, we introduce two techniques for selecting substitution waveforms. One method is based on pattern matching and the other technique explicitly estimates voicing and pitch. Both approaches achieve substantial improvements in speech quality relative to silence substitution.

Error correcting codes have two opposite effects on the efficiency of cellular mobile-radio systems. Although they increase the bandwidth per channel, the codes also make signals more robust and thereby reduce the required distance between users of the same frequency band. This paper reports a mathematical study of the interactions of these two effects in determining the efficiencies of a large number of sourcecode and channel-code combinations. Within a statistical performance objective (basebandSNR \geq 11dB for 90 percent of the users), the most efficient scheme in this study has an embedded differential pulse code modulation (DPCM) source code and a rate 1/2 channel code that protects 2 bits of each 4 bit DPCM code word. Based on a conservative model of cellular transmission, we estimate that the bandwidth efficiency is 3 users/cell/MHz of system bandwidth. By contrast, there are only 1.2 users/cell/MHz with uncoded transmission and 4.5 users/cell/MHz with a rather complicated variable-rate scheme. We also observe that the embedded source code, originally devised for variable-rate operation, has a higher baseband SNR than conventional DPCM in certain fixed-rate environments.

Because of the diversity of testing methods, it is not possible, in general, to compare speech quality measurements from different laboratories. A recent experiment had the aim of determining whether comparable results can be obtained when the same test is performed in several different countries. Speech samples in the native languages of Britain, Canada, France, Italy, Japan, Norway and the United States were processed by 38 communications circuits at Bell Laboratories. Recordings of the processed speech were returned to each location for evaluation, on a five-point scale, by native listeners. The communications circuits included 22 references and 16 adaptive differential pcm (adpcm) test circuits. An important part of the data analysis was estimation of the quality of the 16 adpcm conditions at one location given measurements of adpcm quality elsewhere. The results indicate that a reasonably accurate estimate of adpcm quality at location B is the quality measured at A adjusted by an additive constant. The accuracy of this estimate is equivalent to that of a subjective test at location B with about seven evaluations of each transmission condition.

Because of the diversity of subjective testing methods, it is not possible, in general, to compare speech quality measurements from different laboratories. A recent experiment had the aim of determining whether comparable results can be obtained when the same test is performed in several different countries. Tape recordings of speech samples in the native languages of Britain, Canada, France, Italy, Japan, Norway, and the United States were processed by 38 communications circuits in the Communications Methods Research Department of Bell Laboratories. Recordings of the processed speech were returned to each location for evaluation, on a five-point scale, by native listeners. The communications circuits included white noise and speechmodulated noise references, pulse code modulation (PCM) references and four adaptive differential PCM (ADPCM) test circuits, with bit rates between 16 and 48 kbits/s, each with four binary error rates. An important part of the data analysis was estimation of the quality of the 16 ADPCM conditions at locationB, given measurements of ADPCM quality at locationAand knowledge of reference-circuit quality measured atAand atB. The subjective testing data indicate that a reasonably accurate estimate of ADPCM quality at locationBis the quality measured atAadjusted by an additive constant which is the difference between average reference-circuit measurements atBand atA. The accuracy of this estimate is equivalent to that of a subjective test at locationBwith about seven evaluations of each transmission condition. Three physical measurements are less reliable guides to subjective quality. They lead to estimates with less accuracy than the average of three subjective ratings per circuit.

We describe a digital mobile radio system in which users experiencing poor transmission reduce the number of bits per second used for speech coding from 32000 to 28000. They do so in a manner that greatly reduces the binary error rate. The reduced error rate more than compensates for increased quantization noise. We predict two statistics of user satisfaction in a mobile radio service area: POW, the percentage of users experiencing poor or worse transmission, and GOB, the percentage good or better. In a typical environment the maximum number of users that can operate within a 20 MHz band with POW ≤ 5 percent is 137, compared to 91 users when everyone transmits 32 kbits/s. The modulation technique is FH-FSK (frequency hopped, frequency shift keying) and the digital code is embedded ADPCM (adaptive differential pulse code modulation), in the mobile radio service area, signal strength depends on distance from mobile unit to a central base station. There is slow (shadow) fading due to terrain and building variations and rapid (Rayleigh) fading due to multipath propagation. To predict user satisfaction we have combined three different analyses: 1) a theory of FH-FSK, 2) a theory of the signal-to-noise ratio statistics in mobile radio service areas, and 3) a model, based on computer simulations and subjective testing data, of the dependence of subjective speech quality on ADPCM transmission rate and binary error rate.

Within the bit stream of an embedded digital code is a stream that can be decoded to produce a reasonable replica of the analog source signal. Unlike pulse code modulation (PCM), differential PCM (DPCM), is not an embedded code. If C bits/sample are delected from the bit stream of a DPCM encoder with E bits/sample, the decoded analog signal is substantially noisier than the output of a DPCM codec with D = E - C bits/sample. The penalty is 4-10 dB in signal-to-noise ratio (snr). However, with minor modifications to the encoder and decoder, DPCM becomes an embedded code. Embedded DPCM with E bits/ sample at the encoder and D bits/sample transmitted produces exactly the same output as embedded DPCM with D bits/sample encoding and perfect transmission. The snr of embedded DPCM is slightly lower than the Snr of DPCM. The penalty is 0.5-0.8 dB if the minimum transmitted bit rate is 2 bits/sample. It is less than 0.3 dB if D is at least 3 bits/sample. Combined with an appropriate adaptive quantizer the embedded DPCM codec produces embedded ADPCM (adaptive DPCM) for variable rate transmission ranging from 2 bits/sample up to any desired maximum. Applications exist in speech interpolation, packet switching, and hardware architecture.

A two-band coding system has been constructed for the purpose of providing commentary grade (7 kHz bandwidth) speech or music transmission at 56 or 64 kbits/s. The lower band, 0-to-3650 Hz, is coded with 4 bit ADPCM and the upper band, 3600-to-6800 Hz, is coded with 3 bit or 4 bit/sample APCM. The quality of the coded signal makes the method useful for news and sports broadcasts, and possibly for AM remote music broadcasting. The audio sounds better than that produced by two conventional alternatives: 3200 Hz bandwidth with 8-bit/sample coding and 7000 Hz bandwidth with a single 4-bit/sample coder. The sample may be used in any place with access to a 56 kbit/s Dataphone Digital Service port or to other 56 or 64 kbit/s lines. The power consumption is approximately 12 W in the present form; it could be reduced by a factor of at least two by hardware optimization.

A two-band coding system has been constructed for the purpose of providing commentary grade (7 kHz bandwidth) speech or music transmission at 56 or 64 kb/s. The lower band, 0 to 3650 Hz, is coded with 4 bit adpcm and the upper band, 3600 - 6800 Hz, is coded with 3 bit or 4 bit apcm. The quality of the coded signal makes the method useful for news and sports broadcasts, and possibly for AM remote music broadcasting. The audio sounds better than that produced by two conventional alternatives: 3200 Hz bandwidth with 8-bit coding and 7000 Hz bandwidth with a single 4-bit coder. The system may be used in any place with access to a 56 kb/s Dataphone Digital Service port or to other 56 or 64 kb/s lines. The power consumption is approximately 12 watts in the present form; it could be reduced by a factor of at least 2 by hardware optimization.

It is possible to detect the presence of transmission errors in the decoded sample sequence of a differential pcm system. The detection procedure is based on the sample-to-sample differences in the received signal. It infers an error whenever an individual difference is substantially greater than the rms difference. On the detection of an error, the sample affected is replaced by the result of a smoothing procedure and subsequent samples are adjusted to compensate for the error propagation inherent in differential pcm. This difference detection and correction (DDC) scheme substantially improves system performance at high error rates.

In pulse code modulation (PCM) of speech, transmission errors cause changes in the statistical properties of the received samples. Recently published work describes computer simulations of a difference detection and correction (DDC) system which detect errors on the basis of these changes. The system examines the differences between adjacent received samples. If a particular difference exceeds a threshold, which is dependent on the rms value of a block of 64 differences, the quantized sample responsible for this large difference is deemed to be erroneous. A nonlinear filter of the median type is then introduced as a corrector. We now present a comprehensive description of the hardware realization of this system, and present measured results of signal-to-noise ratio (snr) as a function of bit error rate, input power and the error criterion for narrow-band white noise input signals. An improvement of 6 dB in snr is achieved for error rates between 0.2 and 2.0%. The snr results are consistent with subjective impressions of the quality of speech processed by the DDC system.

An experiment has been performed to study the perceptual characteristics of speech processed by ADPCM. We created 18 three-bit and four-bit coders spanning a wide range of quantizer adaptation parameters. Subjects judged the difference between each pair of coders and rated the quality of each coder individually. The difference data reveal three important perceptual dimensions (overall clarity, signal vs. background distortion, muffled vs. hoarse) which are related to various objective measures of coder performance. Overall subjective quality is well predicted by segmental SNR and even better by a linear combination of measures of granular distortion and overload distortion.

We describe an adaptive coding system capable of representing audio frequency signals in a wide variety of digital code formats. To provide adaptive quantization, the step size is adjusted by voltage controlled amplifiers operating with analog control signals. Sine wave signal-to-noise ratio measurements demonstrate important relationships among the main code classifications: pcm, apcm, dpcm, adpcm. The multipurpose coder has proved a useful laboratory aid in the design of special purpose coders. After adjustment and evaluation of parameters using the multipurpose coder one proceeds to an economical, specialized design.

An experiment was performed to investigate: (a) the influence of PCM code parameters on subjective speech quality, (b) objective measures for predicting perceived distortions and (c) optimum combinations of code parameters. The results indicate that listener opinions depended strongly on coder clipping level and step size but only weakly on bandwidth. Clipping noise power proved a poor predictor of perceived overload distortion; clipping percentage was more useful. Granular noise power was a good predictor of granular distortion. For a given bit rate, the coder with the highest quality rating was not the one with minimum total (clipping + granular) noise power, contrary to traditional wisdom.

A popular method of time-varying quantization involves a multlplicative change in quantizer overload point after each sample time. Although this technique produces an average loading factor that is invariant with rms input, it is vulnerable to transmission errors. Here, we introduce a related quantizer which has the advantage of disipating the effects of transmission errors at any desired rate. On the other hand, adaptation is not perfect. The loading factor now depends on rms input and in designing the quantizer one must compromise between the goal of constant loading factor and the goal of rapid error dissipation. We provide a design method and apply it to the modification of a quantizer currently used for speech. The new quantizer dissipates transmission errors with a 5 ms time constant while maintaining accurate adaptation over a 40 dB input range.

In an adaptive quantizer that has been used for speech encoding, the entire amplitude range expands or contracts by a multiplicative constant after each input sample. The constant Midepends only on the magnitude of the current quantizer output. Assuming independent identically distributed input samples, we show that the sequence of quantizer ranges is a stochastically stable process. Furthermore, we derive a key design equation,\Sigma p_{i}xlogM_{i} = O, wherep_{i}(x)is the probability that an input sample is in theith magnitude interval when the ratio of quantizer range to rms signal level isx. A designer may specifyxand solve this equation for multipliers that provide the desired steady-state performance. There are many such sets of multipliers and we show that the adaptation time constant associated with each set decreases as the ratio of the largest multiplier to the smallest multiplier is increased. On the other hand, the spread of the steady-state range distribution about the operating point can be made as small as desired by making this ratio sufficiently small. A bound is obtained for the tradeoff between responsiveness to changing input level and steady-state range accuracy.

A quantizer that must accommodate a wide range of signal powers would ordinarily require a large number of output levels compared to a quantizer of signals with known power. In this paper a quantizlng scheme is described for situations in which the signal power is unknown a priori but remains constant for the duration of a communication. The required number of levels is reduced because the quantizer amplitude range is adjusted during an initial training period to the value appropriate to the current signal power. At each sample time during the training period, the amplitude range is adjusted by a multiplicative quantity that depends on the most recent quantized output and upon the time elapsed since the onset of training. Assuming independent identically distributed inputs, it is shown that the quantizer range converges to a fixed multiple of the rms value of the input. The ratio of final range to rms input is a unique function of the multipliers, and a formula is presented that allows a designer to choose multipliers that result in any desired range-to-signal ratio. For each final ratio, there are many sets of multipliers, and formulas are derived that indicate, for a limited training period, the tradeoff between the accuracy of the quantizer range and the size of the set of signal powers to which the quantizer can adapt. Computer simulations illustrate the effects of particular multiplier sequences.