An Algorithm for Simple Differential Speech Coding Based on Backward Adaptation Technique

Tomić, Stefan S.; Perić, Zoran H.; Nikolić, Jelena R.

doi:10.15388/Informatica.2018.180

Informatica

An Algorithm for Simple Differential Speech Coding Based on Backward Adaptation Technique

Volume 29, Issue 3 (2018), pp. 539–553

Stefan S. Tomić Zoran H. Perić Jelena R. Nikolić

https://doi.org/10.15388/Informatica.2018.180

Pub. online: 1 January 2018 Type: Research Article

Open Access

Received
1 June 2017

Accepted
1 May 2018

Published
1 January 2018

Abstract

This paper presents a simple differential speech signal coding algorithm, based on backward adaptation. The considered algorithm is executed in frame by frame manner, by implementing predictive and adaptive quantization techniques. Both prediction and adaptation are performed backward, based on the previously quantized input signal frame. This enables us to obtain high quality output signal, without increasing the bit rate. This research puts emphasis on the quantizer design, with the optimal support limit determination, and theoretical performance evaluation. Objective quality of the output signal is evaluated through signal to quantization noise ratio (SQNR). We perform theoretical and experimental analysis of the algorithm performance and provide comparative results of implementing speech signal coding techniques with similar complexity. Experimental results show that our simple differential speech coding algorithm satisfies the G.712 Recommendation for high-quality speech coding at the bit rate of 6 bits per sample. This indicates that the algorithm can be successfully implemented in high quality speech signal coding.

References

Chu, W.C. (2003). Speech Coding Algorithms. John Wiley & Sons, New Jersey.

Dinčić, M., Perić, Z., Jovanović, A. (2016). New coding algorithm based on variable-length codewords for piecewise uniform quantizers. Informatica, 27(3), 527–548.

Fang, J., Li, H. (2008). Distributed adaptive quantization for wireless sensor networks: From delta modulation to maximum likelihood. IEEE Transactions on Signal Processing, 56(10), 5246–5257.

Gibson, J.D. (2016). Speech compression. Information, 7(2), 7–23.

ITU-T, Recommendation G.711 (1972). Pulse Code Modulation (PCM) of Voice Frequencies.

ITU-T, Recommendation G.712 (2001). Transmission Performance Characteristics of Pulse Code Modulation Channels.

ITU-T, Recommendation G.711.1 (2008). Wideband Embedded Extension for ITU-T G.711 Pulse Code Modulation.

Jayant, N.S., Noll, P. (1984). Digital Coding of Waveforms. Prentice-Hall, New Jersey.

Kondoz, A.M. (2004). Digital Speech: Coding for Low Bit Rate Communication Systems. John Wiley & Sons, New Jersey.

Makhoul, J. (1975). Linear prediction: a tutorial review. Proceedings of the IEEE, 63(4), 561–580.

Markel, J.D., Gray, A.J. (2013). Linear Prediction of Speech. Springer Science & Business Media, New York.

Na, S. (2004). On the support of fixed-rate minimum mean-squared error scalar quantizers for a Laplacian source. IEEE Transactions on Information Theory, 50(5), 937–944.

Na, S., Neuhoff, D.L. (2001). On the support of MSE-optimal, fixed-rate, scalar quantizers. IEEE Transactions on Information Theory, 47(7), 2972–2982.

Ortega, A., Ramchandran, K. (1995). Forward-adaptive quantization with optimal overhead cost for image and video coding with applications to MPEG video coders. In: IS&T/SPIE’s Symposium on Electronic Imaging: Science & Technology. International Society for Optics and Photonics, pp. 129–138.

Peric, Z., Denic, D., Nikolic, J., Jocic, A., Jovanovic, A. (2013a). DPCM quantizer adaptation method for efficient ECG signal compression. Journal of Communications Technology and Electronics, 58(12), 1241–1250.

Peric, Z., Aleksic, D., Stefanovic, M., Nikolic, J. (2013b). New approach to support region determination of the μ-law quantizer. Elektronika ir Elektrotechnika, 19(8), 111–114.

Perić, Z.H., Nikolić, J.R., Mosić, A.V., Petković, M.D. (2013c). Design of fixed and adaptive companding quantizer with variable-length codeword for memoryless Gaussian source. Informatica, 24(1), 71–86.

Rabiner, L.R., Schafer, R.W. (1978). Digital Processing of Speech Signals. Prentice-Hall, New Jersey.

Ramalho, L., Fonseca, M.N., Klautau, A., Lu, C., Berg, M., Trojer, E., Höst, S. (2017). An LPC-based fronthaul compression scheme. IEEE Communications Letters, 21(2), 318–321.

Ramalho, L., Freire, I., Lu, C., Berg, M., Klautau, A. (2018). Improved LPC-based fronthaul compression with high rate adaptation resolution. IEEE Communications Letters (accepted for publication).

Sarade, S. (2017). Speech compression by using adaptive differential pulse code modulation (ADPCM) technique with microcontroller. Journal of Electronics and Communication Systems, 2(3), 1–9.

Sayood, K. (2017). Introduction to Data Compression. Morgan Kaufmann.

Suma, M.N. (2012). Performance analysis of DPCM and ADPCM. In: International Conference on Electronic Design and Signal Processing (ICEDSP), pp. 19–23.

Tancic, M., Peric, Z., Tomic, S., Simic, N. (2016). Speech signal coding using forward adaptive quantization and simple transform coding. Elektronika ir Elektrotechnika, 22(3), 74–77.

Uddin, S., Ansari, I.R., Naaz, S. (2016). Low bit rate speech coding using differential pulse code modulation. Advances in Research, SCIENCEDOMAIN International, 8(3), 2348–2394.

Zschunke, W. (1977). DPCM picture coding with adaptive prediction. IEEE Transactions on Communications, 25(11), 1295–1302.

Biographies

Tomić Stefan S.

stefan@elfak.rs

S.S. Tomić was born in Vranje, Serbia, in 1989. He received MS degree in electrical engineering from the Faculty of Electronic Engineering, Nis, Serbia, in 2013. He is currently a student of doctoral studies at the same faculty. His current research interests include audio and speech signal coding. He has published over 10 papers, from which 6 are in peer-reviewed international journals.

Perić Zoran H.

zoran.peric@elfak.ni.ac.rs

Z.H. Perić was born in Niš, Serbia, in 1964. He received the BS, MS and PhD degrees from the Faculty of Electronic Engineering, University of Niš, Serbia, in 1989, 1994 and 1999, respectively. He is a full-time professor at Department of Telecommunications, Faculty of Electronic Engineering, University of Niš. His current research interests include the information theory and signal processing. He is an author and co-author of over 240 papers. Dr. Perić has been a reviewer in a number of journals, including IEEE Transactions on Information Theory, IEEE Transactions on Signal Processing, IEEE Transactions on Communications, Compel, Informatica, Information Technology and Control, Expert Systems with Applications and Digital Signal Processing.

Nikolić Jelena R.

jelena.nikolic@elfak.ni.ac.rs

J.R. Nikolić was born in Prokuplje, Serbia, in 1978. She received the BS, MS and PhD degrees in electrical engineering from the Faculty of Electronic Engineering, University of Niš, Serbia in 2003, 2006 and 2011, respectively. She is currently an assistant professor at the Department of Telecommunications at the same Faculty. Her current research interests include audio, speech and ECG signal compression and approximation methods with application in source coding. She has published over 100 papers, from which 41 in SCI journals. She is a member of the Editorial Board of the Journal of Advanced Computer Science & Technology.

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Keywords

speech coding algorithm backward adaptive technique prediction coding differential coding

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