A sufficient condition for the existence of tandem distortion accumulation is described. The distortion for a sample time at which the sufficient condition is satisfied is studied. It is shown that the transcoding distortion is greater than the adaptive differential pulse code modulation (ADPCM) distortion when synchronous code correction is not used and less than the ADPCM distortion when synchronous code correction is used. These results are combined to show that when synchronous code correction is used, the upper bound of the single transcoding distortion is determined by the single transcoding distortion when synchronous code correction is not used. This bound cannot be generalized for multiple tandem cases.>

Introduces the concept of multiply occupied cells which occur in the concatenation of codecs. The existence of a single multiply occupied cell of cardinality two is shown to imply that an alternative codec exists that results in lower end-to-end distortion and that eliminates the multiply occupied cell. The results are extended to the general case of multiple multiply occupied cells of any cardinality.>

This paper details the development of a single chip VLSI processor which provides fully compatible CCITT 32kbps ADPCM transcoding. The chip is designed for multi-channel systems and is capable of processing eight independent channels in a 125 microsecond frame. To achieve this processing power, a specialized microcoded architecture for the ADPCM algorithm is implemented using a full custom two micron VLSI design methodology.

This paper describes the design and implementation of a phase-reversal, tone disabler for a multichannel digital echo canceller. The algorithm employs a fixed sample delay unit, an adaptive phase corrector, two low-pass filters for energy estimation and a simple phase discriminator. The algorithm has been implemented in a multichannel form using the NEC 7720 DSP chip. Results from both computer simulations and real-time implementation are presented for both noise-free and noisy conditions.

This paper presents a real time implementation of a speech coder at 16 kbs. The compression algorithm is a combination of two previously reported techniques: Time Domain Harmonic Scaling and Adaptive Residual Coding. Several modifications are made to reduce the hardware complexity with minimal loss in speech quality, even in the presence of channel errors at a BER of 10-3. A single-board, real-time, full-duplex implementation is achieved using an NEC7720 signal processing chip and an 8085 microcomputer for transmission over an RS-423 digital interface.

This paper describes the study of an approach to speech digitization at mediumband bit rates of 9.6 kb/s to 16 kb/s for the noisy channels. The technique is based on a combination of Time Domain Harmonic Scaling and an Adaptive Residual Coding. The use of fixed wordsize codewords and the absence of any side information transmission makes the system structure quite simple. Simulation studies have shown that this technique produces communication quality speech at 9.6 kb/s and excellent quality speech at 16 kb/s with a channel bit-error-rate as high as 1%.

Variable length source coding schemes offer substantial improvements in data rate for a wide variety of data compression techniques. Unfortunately, practical considerations have limited their use. This paper describes several techniques which allow these powerful codes to be effectively applied. In particular, the problem of loss of code word synchronization due to channel errors is addressed. It is shown how self-synchronizing codes limit error propagation. Also, variable delay and the associated difficulties of buffer synchronization and buffer overflow are considered. The variable delay at the transmitter can be coupled with a variable delay at the receiver to give a fixed system delay. Buffer control then becomes the allocation of this delay between transmitter and receiver. Finally, a switching property of variable length codes to better match the bimodal operation of the source is presented.

This paper describes the study of a new approach to speech digitization at mediumband bit rates of 9.6 to 16 Kb/s. The technique is based on a combination of Time Domain Harmonic Scaling and an Adaptive Residual Coder. Computer simulation studies have shown that this technique is able to produce excellent quality speech at the bit rates in question.

A simplified combination speech codec (coder/ decoder) and filter structure is described. It uses adaptive delta modulation followed by digital processing to produce a filtered log-PCM representation of a speech signal. The simplification results from combining the delta modulation step-size expansion, integration and low-pass filtering into one operation. All multiplications are eliminated and replaced by one-bit shift-and-add operations. The resulting structure is amenable to a single chip implementation using conventional, all-digital technologies while satisfying established filter standards.

Due to the increasing sophisitication and decreasing cost of digital hardware, an all-digital speech communication network is being considered for implementation. Subject to bandwidth limitations, present plans call for both 16 Kilobit Continuously Variable Slope Delta Modulation (CVSD) as well as 2.4 Kilobit Linear Predictive Coding (LPC) terminals. Hence the conversion of CVSD to LPC and vice versa arises naturally in this environment. This paper will focus on the CVSD to LPC conversion problem. After a brief discussion of the component system environments, a structure for the format conversion system will be proposed. Since previous work has shown that pitch and voicing decisions can be made on the CVSD observations, the presentation will focus on the identification of LPC predictor coefficients from the noisily observed CVSD data. The remainder of the paper centers on this and the coefficient correction problem, and a new class of linear estimators is shown to yield conditionally unbiased estimates of the LPC coefficients in both noisy and noiseless environments.

A speech coding algorithm for digital transmission of speech at a rate of 9600 bits per second which can be implemented on a speech processing system is described. The algorithm combines the following: a pitch extraction loop, a pitch compensating adaptive quantizer, a sequentially adaptive linear predictor, an adaptive source coding, and a multipath tree searching to generate very high quality speech output. Although each of these elements has been previously applied to speech coding, the combination of all five of these elements has not been discussed before. Preliminary simulation studies of the algorithm have yielded a signal-to-noise ratio which is indicative of very high quality speech.

A new adaptive quantizer for speech digitization has been derived. It is similar to known adaptive quantizers in that it adjusts its dynamic range to match that of the speech waveform. In addition, it further adjusts its range to compensate for the increased signal strength that follows a pitch pulse. The new quantizer bases its adaptation on its own output and no side information is required. When combined with a variable length source coding scheme, the new quantizer offers a significant improvement in signal-to-noise ratio and in subjective speech quality. The technique is applicable to a broad range of digitization methods including adaptive delta modulation and various forms of ADPCM.

An improved system for speech digitization using adaptive differential pulse-code modulation (ADPCM) is described. The system uses an adaptive predictor, an adaptive quantizer, and a variable length source coding scheme to achieve a 4-5 dB increase in signal-to-noise ratio over previous ADPCM. The increase can be used to improve speech quality at moderate data rates on the order of 16 kbits/s or to retain the same quality and reduce the data rate to 9.6 kbits/s. The latter alternative permits the use of narrow-band channels. The implementation complexity is on the same order as other ADPCM systems.

In this correspondence, it is shown that an adaptive quantizer with a one word memory can be viewed as one that estimates the variance of its input and normalizes the input by the square root of the estimate. It is shown that, even though the estimate is an exponential average, the effect of transmission errors does not die out. Finally, a method of combating the effect of such errors is described.

A new method of speech digitization called residual encoding is introduced, and its application to the speech digitization problem is studied. The residual encoding system is a form of differential pulse code modulation which utilizes both an adaptive quantizer and an adaptive predictor. The residual encoder differs from previous systems in two ways. First, a sequential estimation method is used to continuously update the predictor coefficients, and second, the predictor coefficients are not transmitted, but are extracted from the estimate of the speech signal at both the transmitter and receiver. No form of pitch extraction is employed. The residual encoding system with a Kalman filter or a stochastic approximation algorithm for identifying the predictor coefficients has produced good quality speech at a data rate of 16 kbit/s.

This paper explores some new solutions to the problem of high-speed data reconstruction in real time. Two types of nonresetting data reconstructors are developed. Both types are simpler than the classical extrapolative hold circuits and when optimized give better performance. The optimization procedure is given and results of simulation studies are included.

Modern digital computing systems are primarily limited in how densely information may be recorded on the magnetic storage elements by the manner in which the information is retrieved (detected) from these storage facilities. Current systems store information at a density around 1000 bits per inch (bit/in); system elements, other than the detection process, could easily handle densities in excess of 3000 bit/in. This paper describes the signal present at the output of the read-write head and suggests two systems by which detection at high bit densities may be accomplished. The error rate associated with each system is given and the results are applied to a computer system using a magnetic disk file as the storage element. It is demonstrated that these detection procedures can meet the widely accepted standard of one error in one billion bits at reasonable signal-to-noise ratios.