Hybrid Companding Delta Modulation

We present a reasonably complete account of an improved adaptive delta modulation (ADM) system called hybrid companding delta modulation (HCDM). The HCDM system that is far superior to continuously variable slope DM (CVSD) or constant factor DM (CFDM) is advantageous, particularly for speech coding. It employs both syllabic and instantaneous companding schemes. Performance analysis of the system has been done and verified by computer simulation. In getting the mathematical formula for HCDM granular noise, a new method based on amplitude distribution is proposed. Optimization of the system parameter values by simulation is also discussed. In addition, an efficient method of hardware implementation is considered.     

Design and simulation of an efficient adaptive delta modulationembedded coder

An embedded coding version of hybrid companding delta modulation (HCDM) is described that operates from 16 to 48 kb/s in 8 kb/s steps. The embedded HCDM coder employs the explicit noise coding technique to transmit an adaptive PCM (APCM) coded version of the HCDM reconstruction error signal as a supplementary bit stream that may be partly or wholly deleted in transmission. SNR performance with speech input depends critically on the design of the supplemental APCM code and two new coding algorithms are investigated. In algorithm 1, the basic cue for step size adaptation is obtained from the RMS slope energy of the HCDM output whereas in algorithm 2, the HCDM reconstruction error is logarithmically compressed before quantisation and the basic step size is derived from peak input magnitudes. Instantaneous adaptation for both algorithms is achieved by using step size multipliers which are optimised for operation at single fixed bit rates and also for decoding with an unknown number of input bit deletions. Simulation results show that SNR performance is significantly enhanced using either algorithm and a graceful reduction of reconstructed speech quality with progressive bit deletion is achieved over the range from 48 kb/s to 16 kb/s. On the whole, the SNR performance of the embedded HCDM system is superior in comparison with conventional HCDM     

Hybrid delta modulators for digital encoding at low bit rates

A performance comparison study of four HDM systems is presented. The four systems studied are hybrid companding delta modulation (HCDM), hybrid constant factor incremental DM (HCF1DM), song hybrid companding DM (SHCDM) and modified SHCDM (MSHCDM). The study was done at low bit rates with three different input signals. According to our results, the MSHCDM always performs better while the HCDM systems may not always be good. Further, for low frequency input signals the dynamic range of SHCDM and HCFIDM is much wider than that of HCDM. In a noisy channel, the results for HCDM at a bit rate of 24 kbs^-1 indicate that the values of SNR at an error rate of 0·001 and 0·01 are, respectively, 15 dB and 6 dB, whereas in the case of HCFIDM and SHCDM they are 16 dB and 10 dB.     

A Study of the Comparative Performance of Adaptive Delta Modulation Systems

A performance comparison of three representative ADM systems has been made by computer simulation using real speech. The three systems studied are continuously variable slope delta modulation (CVSD), Jayant's constant factor delta modulation (CFDM), and a modified version of Un and Magill's hybrid companding delta modulation (HCDM). Among the three systems, HCDM yields the best performance in signal-to-quantization noise ratio (SQNR) and dynamic range regardless of the channel bit error rate. Comparing CVSD and CFDM in an ideal channel, the dynamic range of the latter is significantly wider than that of the former, although their peak SQNR's are almost the same. In a noisy channel, CFDM degrades more rapidly than the other two as the bit error rate increases. In the channel with an error rate above 10^-3, the use of CFDM appears to be impractical when the bit rate is below 16 khits/s. However, intelligible speech transmission is possible with HCDM or CVSD even at the error rate of as high as 10^-1.     

Hybrid Companding Delta Modulation with Variable-Rate Sampling

Hybrid companding delta modulation (HCDM) is known to be superior in performance to other instantaneous or syllabic companding delta modulation systems [1]. To improve its performance or to reduce the bit rate further in coding speech, we propose to use a variable-rate sampling scheme in the HCDM system. The proposed system employs several different sampling rates but transmits the output binary signal at a fixed rate using a buffer. By using the variable-rate scheme, one can improve its performance by 3 to 4 dB in signal-to-quantization noise ratio (SQNR) over the fixedrate HCDM. Detailed algorithm and computer simulation results are presented. Buffer behavior and its control are also discussed. In addition, it is shown that the performance gain of a DM system with variable-rate sampling depends on the degree of variation of the input signal.     

A Multisubscriber Variable-Rate Sampling HCDM System with Dynamic Buffer Control

In this paper we present a multisubscriber variable-rate sampling hybrid companding delta modulation (HCDM) system for simultaneous transmission of several speech signals. This system employs both the statistical multiplexing and variable-rate sampling schemes. It transmits speech signals synchronously at a fixed rate using a buffer. In this system the sampling rate of each subscriber is varied according to the speech activity and the status of buffer occupancy, and only the speech portion is coded for transmission. To optimize the system performance within the allowed maximum transmission delay (300 ms), an efficient dynamic buffer control algorithm is proposed. When the number of subscribers is six and the transmission rate for each subscriber is 16 kbits/s, the proposed system yields a performance improvement of about 10 dB over the conventional single-subscriber HCDM system. The buffer delay in this case is 150 ms, which gives a perceptually negligible effect.     

Implementation of a Multirate Speech Digitizer

In this paper, implementation of a compact and efficient multirate speech digitizer with variable transmission rates of 2.4, 4.8, 9.6, and 14.96 kbits/s is presented. The multirate algorithm has been made based on the residual-excited linear prediction (RELP) vocoder with a transmission rate of 9.6 kbits/s. The residual encoder employed in the RELP vocoder uses hybrid companding delta modulation (HCDM). This HCDM is also used as a 14.96 kbit/s coder. If the residual in the RELP system is down-sampled before encoding, a 4.8 kbit/s coder can be realized. If the residual encoder is not used, a 2.4 kbit/s linear predictive coder (LPC) can be realized by incorporating a pitch extractor. In the 4.8 and 9.6 kbit/s coders the pitch-implanted residual excitation method has been used to generate the excitation signal to the synthesis filter. The multirate speech digitizer algorithm has been implemented using 2900 series bit-slice microprocessors. The external memory is composed of 2K RAM's and 2K ROM's. The system design is a two-bus structure with a 204 ns cycle time. With efficient hardware and software design, the multirate speech digitizer requires almost the same hardware complexity as compared with the conventional 2.4 kblt/s LPC vocoder.     

Digital Companding Techniques

This paper deals with the requirements for the design of digital companding techniques in either delta or pulse-code modulation. Both delta and pulse-code modulation convert analogue signals into binary signals and in both these systems the dynamic range is normally small. By the use of companding, the dynamic range can be extended. Since both delta and pulse-code modulation are digital methods, they are well suited to the use of digital companding techniques. The binary transmitted signal normally contains a measure of the system performance. By observing certain patterns in this binary signal and using the occurrence or nonoccurrence of these patterns to change the gain of the modulator and demodulator, syllabic companding can be obtained. The selection of the binary pattern and the rate of change of gain of the modulator and demodulator, determines both the point at which the companding operates and the attack and decay times. The ratio of the largest to the smallest value of the gain determines the dynamic range. By the use of digital circuitry, the gain can be controlled with sufficient accuracy over a large dynamic range. The paper deals with the principles involved in selecting the binary patterns to control the gain of the modulator and as examples a delta modulation system and a pulse-code modulation system with companding ratios of 60 dB are discussed.     

A hybrid technique for the PAPR reduction of NOMA waveform

Non-orthogonal multiple access (NOMA) is a great contender for future cellular modulation due to its desirable properties like massive connectivity, high data rate transmission, and high spectral efficiency. However, its peak-to-average power ratio (PAPR) is significant, which becomes a significant disadvantage for the efficient operability of the NOMA waveform compared to current techniques. Several PAPR reduction algorithms like selective mapping (SLM), partial transmission sequence (PTS), and companding techniques have been proposed to lower the PAPR of multicarrier waveforms (MCWs). PTS reduces the PAPR but has high complexity. On the other hand, SLM has a less complex framework, but its PAPR performance is not as efficient as PTS. Companding methods reduce the PAPR by compressing the signals at the transmitter, which unfortunately reduces the dynamic range of the signal. In this work, we propose a hybrid algorithm (SLM + PTS) with a companding method for the first time for the NOMA waveform, which efficiently reduces the PAPR with low computational complexity. Furthermore, we compare the performances of a host of candidate algorithms like SLM, PTS, hybrid (SLM + PTS), hybrid + A law (SLM–PTS–A law), and hybrid + Mu law (SLM–PTS–Mu law). The results of the experiments show that the hybrid + Mu law did a better job than the existing PAPR reduction algorithms.     

Exponential companding technique for PAPR reduction in OFDM systems

In this paper, a new nonlinear companding technique, called "exponential companding", is proposed to reduce the high Peak-to-Average Power Ratio (PAPR) of Orthogonal Frequency Division Multiplexing (OFDM) signals. Unlike the /spl mu/-law companding scheme, which enlarges only small signals so that increases the average power, the schemes based on exponential companding technique adjust both large and small signals and can keep the average power at the same level. By transforming the original OFDM signals into uniformly distributed signals (with a specific degree), the exponential companding schemes can effectively reduce PAPR for different modulation formats and sub-carrier sizes. Moreover, many PAPR reduction schemes, such as /spl mu/-law companding scheme, cause spectrum side-lobes generation, but the exponential companding schemes cause less spectrum side-lobes. Computer simulations, which consider a baseband OFDM system with Additive White Gaussian Noise (AWGN) channels and a Solid State Power Amplifier (SSPA), show that the proposed exponential companding schemes can offer better PAPR reduction, Bit Error Rate (BER), and phase error performance than the /spl mu/-law companding scheme.     

Linear and adaptive delta modulation

New results are presented, offering insight into the performance and optimization of linear and adaptive delta modulation, together with a comparison with pulse code modulation. The results are applied to three cases of practical interest: television, speech, and broadband signals. The results are presented as follows: first, a characterization of the quantization noise of linear delta modulation (DM) is given; second, an adaptive DM system which seems promising for television and speech is evaluated; and third, a comparison between PCM and adaptive DM is made for speech, television, and broadband signals. It is concluded that 1) the adaptive system provides DM with a companding capability, 2) adaptive DM offers a bit rate or channel bandwidth reduction capability in comparison with PCM for television signals, 3) adaptive DM appears better suited to television and speech signals than linear DM, 4) the maximum S/N performance of adaptive DM is the same as that of linear DM, 5) the companding improvement offered by adaptive DM is not limited by the same practical considerations as those of PCM, and 6) the S/N performance of adaptive DM is the same for both Gaussian and exponential signal densities.     

IGCC for PAPR Reduction in OFDM Systems Over the Nonlinearity of SSPA and Wireless Fading Channels

High peak-to-average power ratio (PAPR) in orthogonal frequency division multiplexing (OFDM) systems seriously impacts power efficiency in radio frequency section due to the nonlinearity of high-power amplifiers. In this article, an improved gamma correction companding (IGCC) is proposed for PAPR reduction and investigated under multipath fading channels. It is shown that the proposed IGCC provides a significant PAPR reduction while improving power spectral levels and error performances when compared with the previous gamma correction companding. IGCC outperforms existing companding methods when a nonlinear solid-state power amplifier (SSPA) is considered. Additionally, with the introduction of \(\alpha , \beta , \gamma \), and \(\varDelta \) parameters, the improved companding can offer more flexibility in the PAPR reduction and therefore achieves a better trade-off among the PAPR gain, bit error rate (BER), and power spectral density (PSD) performance. Moreover, IGCC improves the BER and PSD performances by minimizing the nonlinear companding distortion. Further, IGCC improves signal-to-noise ratio (SNR) degradation (\(\varDelta _{\mathrm{SNR}}\)) and total degradation performances by 12.2 and 12.8 dB, respectively, considering an SSPA with input power back-off of 3.0 dB. Computer simulation reveals that the performances of IGCC are independent of the modulation schemes and works with arbitrary number of subcarriers (N), while it does not increase computational complexity when compared with the existing companding schemes used for PAPR reduction in OFDM systems.     

降低正交频分复用系统峰均功率比的部分压扩算法

针对现有压扩变换法系统性能差的缺点,提出了一种降低系统峰均功率比的部分压扩算法。该方法根据正交频分复用(OFDM)系统信号幅度服从瑞利分布的统计特性,仅压缩大幅度信号保持了系统信号幅度的分布特征,弥补了现有压扩变换的不足,且具有带外功率小的优点。在M阶调制方式下的系统仿真结果表明,部分压扩方法与选择性映射和部分传输序列等方法相比,可获得相近的峰均功率比压缩效果并且在同样的系统误码率条件下比指数压扩法获得约log2(M)dB信噪比增益。     

Performance characteristics of cellular systems with different link adaptation strategies

In this paper, the theoretical performance of cellular systems with different types of link adaptation is analyzed. A general link and system performance analysis framework is developed to enable the system-level performance characteristics of the various link adaptation strategies to be studied and compared. More specifically, this framework is used to compare the downlink performance of fully loaded cellular systems with fixed power and modulation/coding, adaptive modulation/coding (AMC), adaptive power allocation (APA) with system-level AMC, and water-filling (WF). Performance is studied first for idealized methods, and then for cases where some practical constraints are imposed. Finally, a hybrid link adaptation scheme is introduced and studied. The hybrid scheme is shown to overcome most of the performance loss caused by the practical constraints. Moreover, the hybrid scheme, as opposed to WF, enables the system to be tuned to meet the most important performance objective for the system under consideration, such as coverage reliability, capacity, or data rate distribution. The algorithms and the framework presented in this paper can be used to improve the link adaptation performance of modern cellular systems such as HSDPA.     

An investigation of delta modulation of speech for mobile radio

A characteristic of a mobile radio channel is the occurrence of correlated signal fading that results in burst errors. The use of adaptive delta modulation (ADM) based on explicit transmission of the quantizer step size was proposed earlier for speech communication over such a channel. Two other variable step-size delta modulation (VSDM) schemes are presented, and their performance in a mobile radio environment is discussed. One of them is the constant factor delta modulation that uses one-bit memory and produces fast and instantaneous step-size changes. The other is the digitally controlled delta modulation (DCDM) that incorporates a new step-size adaptation strategy using seven bits of memory. In some cases, bit scrambling has been used. This is equivalent to scrambling (spreading out in time) the, clustered errors. Computer simulations providing values of coder parameters for satisfactory signal-to-noise ratios for band-limited speech signals and Gaussian noise are described. New hardware realizations are given that allow those parameters to vary smoothly for a wide range of sampling frequencies. Results of informal listening tests obtained with a mobile radio channel simulator are included. It is shown that for mobile radio, DCDM, as expected, is the better of the two coders. This is because it does not sacrifice its overload performance for the sake of error resistance.     

Digital coding of speech waveforms: PCM, DPCM, and DM quantizers

A study is presented on the digital coding of speech by means of a straightforward approximation of the time waveform. In particular, the closely related discrete-time discrete-amplitude signal representations that are rather well known as pulse-code modulation (PCM), differential pulse-code modulation (DPCM), and delta modulation (DM) are discussed. Speech is recognized as a nonstationary signal, and emphasis is therefore placed on "companding" and "adaptive" strategies for waveform quantization and prediction. With signal-to-quantization-error ratio SNR as a performance measure, techniques are suggested which are most likely to be appropriate for given specifications of information rate. It is pointed out that error waveforms in speech quantization cannot be regarded as additive white noise, in general. This means that for finer assessments of speech coders, either relative or absolute, one needs to supplement SNR-based observations with corrections for subjective and perceptual factors. The latter seem to defy quantification as a rule. Invaluable, therefore, are explicit preference tests for direct comparisons of coders from a perceptual standpoint, and notions such as isopreference and multidimensional scaling are naturally appropriate in interpreting the results of such tests. Final points of concern are communication questions such as multiple encodings of speech by tandem coder-decoder pairs; conversions among different digital code formats; and the effects of additive and multiplicative noise in the communication channel, as manifest in the erroneous reception of speech-carrying bits. Information on these topics tends to be heterogeneous and nontheoretical, and the present digression into the subject is cursory by intent. The gramophone record accompanying this paper demonstrates some of the manipulations of speech that are discussed.     

Combined power and rate adaptation for wireless cellular systems

We extend the throughput optimization technique of Qiu and Chawla (1999) for adaptive modulation, to combine power and rate adaptation in wireless cellular systems. We develop new combined power and rate control algorithms for wireless multimedia systems, in which the transmitted powers and rates of different media users are adapted based on the signal-to-interference power ratio. Using simulations, we show that with appropriately chosen power and rate limits, our proposed combined power and rate control algorithms can achieve a higher throughput when compared to previously proposed algorithms with power control only.     

On the Nonlinear Companding Transform for Reduction in PAPR of MCM Signals

In this paper, we provide the design criteria of the nonlinear companding transforms for reduction in peak-to-average power ratio (PAPR) of multi-carrier modulation (MCM) signals, which can enable the original MCM signals to be transformed into the desirable distribution. As examples, some novel nonlinear companding transforms have been proposed to transform the amplitude or power of the original MCM signals into uniform distribution, which can effectively reduce the PAPR for different modulation formats and subcarrier sizes without any complexity increase and bandwidth expansion. It has been shown by computer simulations that the proposed schemes can significantly improve the performance of MCM systems including bit-error-rate and PAPR reduction.     

Hybrid Zadoff‐Chu and multilateral piecewise exponential companding transform–based PAPR reduction technique in OFDM systems

The orthogonal frequency‐division multiplexing (OFDM) is a multicarrier modulation system that is used to transmit the large volume of data to the receiver. Reducing the peak‐to‐average power ratio (PAPR) in OFDM system is one of the demanding and crucial task in recent days. For this reason, various precoding and companding mechanisms are developed in the traditional works, but it remains with the limitations of increased complexity, reduced performance, and nonlinear distortion. The reduction of PAPR is achieved by minimizing the companding distortion with the enhancement of the bit error rate (BER) performance significantly. Then, in order to avoid clipping in OFDM, a multilateral piecewise exponential companding transform (MPECT) method has been utilized rather than using piecewise exponential companding transform (PEC) where PAPR is getting reduced. The OFDM is very sensitive to synchronizing error. To overcome this sensitivity, employ the Zadoff‐Chu sequence to carrier frequency offsets. Zadoff‐Chu matrix transform (ZCMT) has numerous merits among the other ODFM systems such as the improvement in the performance of the channels that are fading away and provides an ideal periodic autocorrelation and a constant magnitude periodic cross correlation. Both of these techniques provide improvement in the ODFM systems. To get more efficiency, this paper aims to develop a hybrid technique by integrating the ZCMT and MPECT techniques for reducing the PAPR in OFDM systems. Further, convolutional encoding is applied for better BER and PAPR. The simulation results of the proposed ZCMT‐MPECT technique are evaluated and compared with the conventional OFDM and other precoding methods.     

Linear Companding Transform for the Reduction of Peak-to-Average Power Ratio of OFDM Signals

A major drawback of orthogonal frequency-division multiplexing (OFDM) signals is their high peak-to-average power ratio (PAPR), which causes serious degradation in performance when a nonlinear power amplifier (PA) is used. Companding transform (CT) is a well-known method to reduce PAPR without restrictions on system parameters such as number of subcarriers, frame format and constellation type. Recently, a linear nonsymmetrical companding transform (LNST) that has better performance than logarithmic-based transforms such as $mu$-law companding was proposed. In this paper, a new linear companding transform (LCT) with more design flexibility than LNST is proposed. Computer simulations show that the proposed transform has a better PAPR reduction and bit error rate (BER) performance than LNST with better power spectral density (PSD).