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.     

Two Loop Adaptive Delta Modulation Systems

Two adaptive delta modulation (ADM) coders, one for A/D conversion of speech and the other for fiat spectrum signals, have been developed by incorporating a second adaptive feedback loop around the basic delta modulation (DM) coder. The dynamic range of the coder for integrated spectrum has been extended to more than 35 dB with a signal-to-noise ratio (SNR) > 28 dB, for a pulse-repetition frequency (PRF) of 32 kbits/s only, by using an adaptive circuit consisting of digital/analog slope detector and an FET expander. For flat spectrum signals, similar results with constant SNR over the band are obtained at a bit rate of 40 kbits/s. Thus the coders have performances equivalent to those of a 7-digit PCM, using only 32 and 40 kbits/s, respectively.     

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.     

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.     

A New Channel Bank with Block Companding

Performance improvement of the nearly instantaneous companding PCM (or block companding PCM) is discussed. A simplified algorithm is proposed. The design and performance of a channel bank adopting the proposed algorithm is described. The channel bank can transmit 44 voiceband signals on a standard T1 line. It provides commercially acceptable speech quality as well as sufficient transparency for voiceband data up to 4800 bits/s.     

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.     

Time-Shared Codecs for Multichannel Delta Modulation and Differential PCM

Current practice for multichannel delta modulation (DM) terminals uses one DM codec per channel-end. This paper describes methods by which one high-speed DM codec can be timeshared over a large number of channels. The methods are also applied to multichannel differential PCM (DPCM) terminals. In both cases the time-shared codecs include step-size adaptation determined by the recent past history of the coded digital signal. A method for digital conversions between multichannel linear PCM and adaptive DPCM (ADPCM) formats is also described.     

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     

Optimum Weighted PCM for Speech Signals

Weighted digital modulation schemes which provide bit error probabilities matched to the PCM bits with respect to their sensitivity to digital errors are analyzed. The channel is additive, white Gaussian. The PCM system has arbitrary code, companding law and input signal density function. Especially optimum weighted PSK/PCM and QAM/PCM are given for speech signals. The average channel signal to noise ratio is kept constant when schemes are compared. We obtain a channel signal to noise ratio gain in threshold extension of 2 dB for standard 8 bit PCM. The performance of suboptimum schemes, where the number of different bit error probability levels are smaller than the number of PCM bits are also studied. Two levels per 8 bit PCM word yield more than half of the achievable gain (in dB) and 4 levels is almost equal to optimum.     

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.     

A Per-Channel A/D Converter Having 15-Segment µ-255 Companding

This paper describes a companded analog-to-digital (A/D) converter for voiceband signals that is simple and potentially inexpensive. The converter uses only 18 coarsely spaced analog levels. Fine resolution is obtained by oscillating between these levels at an increased speed and averaging the result over a Nyquist interval. The companding used in the converter is effectively the same as that of μ-255 pulse-code modulation (PCM). In the encoding process a one-bit code is generated at 256 000 samples/s. This 1-bit per sample signal can be transmitted and decoded directly, or a simple digital circuit will produce a 13-bit, 8-kHz linear PCM signal that can be compressed to 8-bit companded PCM format. In this paper the basic operation of the 1-bit coder is described and its performance when connected to a 1-bit decoder is illustrated. Methods for obtaining both linear and compressed PCM are then presented, and the properties of these PCM signals with respect to noise, gain tracking, and harmonic content are described. Relative insensitivity to circuit component variations, absence of analog gates, along with the need to generate only a few analog levels, make the coder especially well suited to integrated circuit realization.     

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.     

A Comparison of Optimum and Logarithmic Quantization for Speech PCM and DPCM Systems

Two quantization rules, optimum (minimum mean-square error) and logarithmic companding, are compared for application to digital speech transmission. Comparison is made on the basis of signal-to-quantizing noise ratio (S/N), subjective quality judgments, idle channel noise, and dynamic range. These quantizers are considered in both PCM and differential PCM configurations. A computer algorithm is described that yields the optimum quantizer levels for a given speech record. It has been found that the improvement in S/N which optimum quantization affords over conventional logarithmic quantization is offset by the greater idle channel noise and smaller dynamic range (range of talker volumes handled with a lower limit on S/N) of the optimum law.     

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.     

Voice-Band Data Modem Performance Over Companded Delta Modulation Channels

This paper reports upon the results of tests of the transmission of data over single- and multiple-hop companded delta modulation (DM) systems. The DM coder-decoder (CODEC) was optimized for voice transmission. Modem bit error rate (BER) achievable over the range of 1200-9600 bits/s is presented. A comparison with the performance of pulse code modulation (PCM) is included. The comparison indicates that the two systems are comparable for error-free digital lines but favors the DM system for lines with errors.     

Differential pulse-code modulation (PCM) with entropy coding

As the transmission rateRgets large, differential pulse-code modulation (PCM) when followed by entropy coding forms a source encoding system which performs within 1.53 dB of Shannon's rate distortion function which bounds the performance of any encoding system with a minimum mean-square error (mmse) fidelity criterion. This is true for any ergodic signal source. Furthermore, this source encoder introduces the same amount of uncertainty as the mmse encoder. The 1.53 dB difference between this encoder and the mmse encoder is perceptually so small that it would probably not be noticed by a human user of a high quality (signal-to-noise ratio(S/N) geq 30dB) speech or television source encoding system.     

An Improved Hybrid Companding Delta Modulator

We present two new forms of "hybrid companding" delta modulation for speech encoding. These algorithms differ from the earlier hybrid companding delta modulation (HCDM) scheme in that they employ Song voice adaptation as the basis of the instantaneous companding, rather than constant factor adaptation. We show that although the two new algorithms produce SQNR curves which are similar to that of HCDM, a subjective evaluation of the three systems significantly favors the new ones. In addition, the new algorithms offer simpler implementation in digital hardware.     

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.     

Realization of ADM Arithmetic Signal Processors

The purpose of this paper is to show that signals which are adaptive delta modulation (ADM) encoded can be arithmetically processed directly, without first decoding or converting to pulse code modulation (PCM) format. By operating on the serial DM bit streams, the sum, difference and product can be obtained in PCM and ADM format. Employing a four-term, non-recursive, averaging filter after the processors, we show that, for constant inputs, the signal-to-noise ratio (SNR) of the DM devices is exactly the same as that of their PCM counterparts. Although we have used the Song audio mode ADM [1] in the realization of our arithmetic processors, the designs are general enough to be applied to a large class of digital ADMs. To keep all systems practically realizable, we only employ operations which can be constructed with standard digital hardware, that is, adder, delays, hard-wired scalars and common logic circuits. Consequently, all our ADM devices can be manufactured with large scale integration where the distinct advantage is low cost and high reliability.     

Digital Encoding of Phase Shift Keying Voiceband Data Signals

Computer simulation was used to evaluate the performance of eleven coder/decoders (CODEC's) with phase shift keying (PSK) and differential PSK(DPSK) voiceband data signals. The CODEC's were PCM, differential PCM and delta modulation systems designed for speech and operating at bit rates from 16 to 64 kbits/s. The voiceband data signals processed by these CODEC's were demodulated to determine the phase error caused by the CODEC. The phase error introduced by the CODEC's is a function of the phase of the CODEC sampling clock relative to the data modem bit clock. Some of the statistics of the phase error are presented. Three performance metrics were used to evaluate the performance of these CODEC's-signal to quantizing noise ratio, variance of the phase error and maximum value of the phase error.