Continuously Variable Slope Delta Modulation Research Articles

An efficient two-digit adaptive delta modulation for Laplacian source coding

ABSTRACTDelta Modulation (DM) is a simple waveform coding algorithm used mostly when timely data delivery is more important than the transmitted data quality. While the implementation of DM is fairly simple and inexpensive, it suffers from several limitations, such as slope overload and granular noise, which can be overcome using Adaptive Delta Modulation (ADM). This paper presents novel 2-digit ADM with six-level quantization using variable-length coding, for encoding the time-varying signals modelled by Laplacian distribution. Two variants of quantizer are employed, distortion-constrained quantizer which is optimally designed for minimal mean-squared error (MSE), and rate-constrained quantizer, which is suboptimal in the minimal MSE sense, but enables minimal loss in SQNR for the target bit rate. Experimental results using real speech signal are provided, indicating that the proposed configuration outperforms the baseline ADM algorithms, including Constant Factor Delta Modulation (CFDM), Continuously Variable Slope Delta Modulation (CVSDM), 2-digit and 2-bit ADM, and operates in a much wider dynamic range.

비균등 표본화 기법과 음성 부호화로의 응용

국소적으로 선형적인 특성을 보이는 음성신호와 같은 신호의 데이터율을 감소시키기 위하여 변곡점 검출에 기반한 비균등 표본화 방법을 제안한다. 국소적인 최대값과 최소값 검출에 기반하여 비균등 표본화를 수행하는 기존의 방법에 비하여 변곡점에 기반한 비균등 표본화는 신호의 기하학적인 특징을 충실히 활용한다. 결과로서, 변곡점 검출에 기반하여 비균등 표본화된 데이터를 보간법으로 처리하면 기존의 방법보다 원시신호를 정밀하게 복원할 수 있다. 컴퓨터 모의실험을 통하여 기존의 최대값/최소값 검출 방법에 비교해서 제안된 변곡점 검출 기반의 비균등 표본화가 약 9~23dB의 신호대 잡음비 개선효과가 있음을 확인하였다. 제안된 변곡점 검출 기반의 비균등 부호화의 유용성을 보이기 위하여 음성신호의 부호화에 적용하였으며, Continuously variable slope delta modulation (CVSD)방법과 성능을 비교하였다. 변곡점 표본은 1로 설정된 플래그와 함께 이진수로 전송되며, 비 변곡점은 플래그만 0으로 설정된다. 음성신호에 따라 약 0.3 ~ 9dB의 신호대 잡음비 개선효과가 있으며, 주관적인 성능지표인 Mean Opinion Score (MOS)는 약 0.5 ~ 1.3 개선되었다. For a signal such as speech showing piece-wise linear shape in a very short time period, a nonuniform sampling method based on the inflection point detection (IPD) is proposed to reduce data rate. The method exploits the geometrical characteristics of signal further than the existing local maxima/minima detection (MMD) based sampling method. As results, the reconstructed signal by the interpolation of the IPD based sampled data resembles the original speech more. Computer simulation shows that the proposed IPD based method produces about 9~23 dB improvement over the existing MMD method. To show the usefulness of the IPD technique, it is applied to speech coding, and compared to the continuously variable slope delta modulation (CVSD). The nonuniformly sampled data is binary coded with one bit flag set "1". Noninflection samples are not sent, but only flag bits set 0 are sent. The method shows 0.3 ~ 9 dB SNR and 0.5 ~ 1.3 mean opinion score (MOS) improvements over the CVSD.

가변 비트율 음성 부호화기의 성능분석

A variable bit rate speech coder is presented. The coder is based on the observation that a speech signal can be viewed as a combination of piecewise linear signals in a short time period. The encoder detects the sample points where the slope of the signal changes, which are called the inflection points in this paper. The coder transmits the location and value for the detected inflection sample, but only the location information for the noninflection samples. In the decoder, the noninflection samples are estimated with interpolation of the received information. Several factors affecting the performance of the coder have been tested through simulation. Simulation results show that the linear interpolation produces 1 ~ 5 dB improvement over the cubic spline interpolation. And the -law companding does not provide any benefit when it is applied before the inflection detection. With low threshold values in the inflection point detection, the coder shows better MOS and more than 16 dB improvement in SNR compared to the continuously variable slope delta modulation (CVSDM).

Single-chip speech recognition system based on 8051 microcontroller core

This paper describes a single-chip speech recognition system. It contains the speech functions of prompt, playback, speaker-dependent speech recognition, suitable for the voice activated systems in toys, games, consumer electronics, office devices, etc. The chip is designed based on the SOC (system on chip) philosophy and an 8-bit MCU, RAM, ROM, ADC/DAC, PWM, I/O ports and other peripheral circuits are all embedded in it. Software modules including control/communication, speech coding and speech recognition algorithms are implemented in an 8051 compatible microcontroller core, resulting in the extremely low cost of the chip. The speech recognition adopts the template matching technique. It recognizes up to 20 phrases with an average length of 1 second and the recognition accuracy reaches more than 95% with the background SNR above 10 dB. Speech coding uses continuous variable slope delta modulation (CVSD) algorithm. The bit rate is 16 kbits/s.

Adaptive line echo cancellation combined withCVSD speech coders

An efficient method for controlling the echo in a continuously variable slope delta modulation (CVSD) mobile communication channel is proposed. By utilising the symmetry of the coder pairs, the proposed method shortens the echo path length and reduces the nonlinear distortion inherent in the coders. An additional advantage for the VLSI implementation is obtained using the combined bit-stream. The proposed method shows an improved echo return loss enhancement (ERLE) performance and reduced complexity compared to the conventional method.

Estimating time-varying noise introduced by CVSD for speech enhancement

The authors examine the possibility of improving the quality of speech obtained with continuously variable slope delta-modulation (CVSD), a speech coding method that has been quite widely employed in mobile radios. CVSD is well-known for its robustness and implementation simplicity, but the quality of the resultant speech (commonly at 16 kbps) is often not very satisfactory. Through their study, the authors obtain two crucial findings on the characteristics of the disturbance/noise introduced by CVSD. Based on their findings, they develop a method for estimating the short-time power spectra of CVSD noise, and this enables them to apply existing speech enhancement methods to effectively suppress CVSD noise. Performance assessments based on objective measures such as signal-to-noise ratio (SNR), segmental SNR, and COSH distortion measure, and informal subjective listening tests have all indicated significant improvements in speech quality.

Character recognition and its Announcing using artificial Neural Networks

This paper presents an Artificial Neural Network based implementation of a new methodology for recognition of the characters and their announcement. The characters identified are announced using pre-recorded voice data. In doing so, the Continuous Variable Slope Delta Modulation (CVSD) has been used. A quick back propagation algorithm has been employed for training the network utilising a sigmoidal nonlinearity. This system will be of great use as a reading aid for the busy and the blind.

1-Bit delayed encoding improves continuously variable slope delta modulator performance

This paper is concerned with the use of 1-bit delayed encoding in the operation of a continuously variable slope delta modulator (CVSDM). Delayed encoding is realized as a time-adaptation process capble of anticipating slope-overload bursts by overriding the ordinary operation of the CVSDM. Practical realization of the delayed CVSDM version shows that its operation offers up to 5 dB S/N improvement for medium sampling rates with no major increase in the encoder's circuit complexity.

A Comparison of Digital Speech Coding Methods for Mobile Radio Systems

Subjective quality measurements on three digital speech coders, simulated with mobile radio channel transmission, were performed using the mean opinion score (MOS) method. The three speech coding methods tested were: continuously variable slope deltamodulation (CVSD) coding, adaptive predictive coding (APC), and residually excited linear predictive (RELP) coding. Several versions of each coder, with transmission rates in the range of 7.3 to 16.1 kbits/s, were simulated. Five different channel conditions, including three derived from land mobile radio field experiments, were applied to the speech coders' encoded output to study the effects. The results show that of the three coders, the CVSD coder is the most robust to channel errors, but produces reconstructed output speech of unacceptable quality. The 14.4 kbit/s RELP coder produces relatively good Output speech quality, exhibits a mild degree of robustness to mobile radio channel errors, and is slightly less complex than the APC coder. Of the three digital speech coders tested, the RELP coder appears the most suitable for use with land mobile radio. However none of the three coders was able to produce speech of telephone toll quality in a mobile radio environment.

Improving the dynamic range of a CVSD coder

The letter describes a method of improving the dynamic range of a continuously variable slope delta modulator (CVSD). This is achieved by modifying the basic step size δ0 Compared to the CVSD algorithm, the modified CVSD (MCVSD) algorithm yields about 15–20 dB dynamic range improvement without degrading the peak SNR and the bit error rate tolerance.

Bit Rate Reduction of Digitized Speech Using Entropy Techniques

We present the results of a study to reduce the bit rate of speech that has been digitized with a continuously variable slope delta modulator (CVSD) operating at 16, 24, and 32 kbits/s. The theoretical reduction is found from the bit stream entropy. The actual reduction, via Huffman coding, is within 1-2 Percent of the theoretical value. The conditional entropy indicates that additional bit rate reduction can be achieved if we use a set of Huffman codes, conditioned on the past CVSD bits. A third technique, tandem coding, using a maximum likelihood predictor in tandem with run length and Huffman coding, is also investigated. Using these entropy techniques, bit rate reductions of 11-25 percent are achieved for the CVSD rates considered. The paper concludes with a study of the buffer requirements needed to support these entropy coders.

Hybrid frequency-domain coding of speech signals

A new method of speech coding called hybrid frequency-domain coding is introduced and its application to speech transmission at bit rates of 7.2 and 9.6 kbit/s is studied. The hybrid frequency-domain coding system is based on a combination of sub-band coding and transform coding techniques. Subjective tests have shown that the quality of speech produced by the hybrid frequency-domain technique at 7.2 kbit/s is comparable with 19 kbit/s continuously variable slope delta modulation (CVSD), and that at 9.6 kbit/s, it is comparable with 23 kbit/s CVSD.

A Performance Study of Delta Modulation Systems for Voiceband Data Signals

In this paper the performance of delta modulation (DM) systems have been studied by computer simulation when the input signal to the DM coder is a voiceband data signal. First, the parameter values of three DM systems, linear DM (LDM), constant factor DM (CFDM), and continuously variable slope DM (CVSD) has been optimized for 4800 bits/s differential phase shift keying (DPSK) signal. Then, the performance of the three DM systems have been studied for ideal and noisy channels. It has been found that the peak signal-to-quantization-noise ratio (SQNR) is nearly the same regardless of coding scheme used, but CFDM yields the widest dynamic range. In a noisy channel, however, CFDM is very sensitive to channel bit errors. Considering the overall performance, CVSD appears to be the best among the three DM systems studied. Also, the performances of DM's have been compared with those of PCM and DPCM systems. In addition, we have studied the effect of DM quantization noise on modem bit error rate by the Monte Carlo simulation method. It is possible to transmit a 4800 bits/s DPSK signal at a bit error rate below 10 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-5</sup> by CVSD with the rate of 32 kbits/s.

Quantization Noise in Adaptive Delta Modulation Systems

New analytical formulas for quantization noise of two widely known adaptive delta modulation (ADM) systems, continuously variable slope delta modulation (CVSD) and Jayant's firstorder constant factor delta modulation (CFDM), have been obtained. These formulas are derived based upon the previously obtained results for linear delta modulation (LDM) and adaptive differential pulse code modulation (ADPCM). To verify the results, computer simulation has been done using band-limited Gaussian signals. Close agreements between the formulas and the simulation results have been obtained in the wide range of the input signal level.

Delta Modulators in Packet Voice Networks

In this paper, the performances of the Song Voice Adaptive Delta Modulator (SVADM) and the Continuously Variable Slope Delta Modulator (CVSD) in terms of dynamic range, sampling rate and the channel errors are compared. The use of the SVADM and the CVSD in a packet voice system, the algorithms for digital detection of silent periods and the performance of a packet voice system using the SVADM and the CVSD as source encoders are presented. The parameters employed for subjective evaluation of the packet voice system are packet size, silence detection algorithm, bit rate and packet loss rate.

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 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-3</sup> , 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 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-1</sup> .

A PCM/ADM and ADM/PCM code converter

In this paper a bidirectional code converter system that converts PCM to ADM and ADM to PCM directly in the digital domain is studied. The particular coding schemes chosen for study are the μ-255 PCM with the rate of 64 kbits/s and the continuously variable slope delta modulation (CVSD) with 32 kbits/s. Efficient realization of the converter hardware is also considered.

Analysis of the performance of a tandem connection of a 2.4-kB/sec LPC vocoder to a 16-kB/sec CVSD coder

The performance of a tandem connection of narrow-band and wide-band speech communication systems is evaluated. Specifically, the narrow-band system consists of a conventional LPC vocoder operating at a bit rate of 2.4 Kb/sec and the wideband system consists of a CVSD (continuously variable slope delta modulator) operating at a bit rate of 16 Kb/sec. This paper describes the properties of the narrowband to wideband link. The SNR (signal-to-quantizing noise ratio) of the CVSD coder is analyzed over a 50-dB variation of the input signal levels and for a variety of source excitations for the LPC synthesizer. It is shown that SNR improvements in the CVSD coder of 2–2.5 dB are possible in the slope overload region of the coder by modifying the source excitation of the LPC synthesizer and by preprocessing the input signal to the coder with an allpass filter. Both methods aid in reducing the peak factor (peak to rms level) of the input speech to the coder. Subjectively, however, only slight improvements in quality, if any, were observed with these modifications. The weakest link in terms of quality in the tandem connection appears to be that of the CVSD coder.