Computationally Efficient Signal Quality Improvement Method for VoIP System
The voice signal in Voice over Internet protocol (VoIP) system is processed through the best effort policy based IP network, which leads to the network degradations including delay, packet loss jitter. The work in this paper presents the implementation of finite impulse response (FIR) filter for voice quality improvement in the VoIP system through distributed arithmetic (DA) algorithm. The VoIP simulations are conducted with AMR-NB 6.70 kbps and G.729a speech coders at different packet loss rates and the performance of the enhanced VoIP signal is evaluated using the perceptual evaluation of speech quality (PESQ) measurement for narrowband signal. The results show reduction in the computational complexity in the system and significant improvement in the quality of the VoIP voice signal.
[1] Goralski, J.W., Kolon, C.M., IP Telephony, 1st ed., McGraw-Hill, 2000.
[2] Varshney, U., Snow, A., Mcgivern, M., Howard, C., "Voice over IP",
Communications of the ACM, 2002, vol.45,no.1, pp.89-96
[3] Verhelst, W., Roelands, M., "An overlap-add technique based on
waveform similarity (WSOLA) for high quality time-scale modification of speech", In Proceedings of IEEE International Conference on
Acoustics, Speech and Signal Processing, 1993, vol.2, pp. 554-557.
[4] Stenger, A., Younes, K.B., Girod, B., Sanneck, H., "A new technique for
audio packet loss concealment", In Proceedings of IEEE Global
Telecommunication Conference, Communications: The key to Global Prosperity, 1996, pp. 48-52.
[5] Feng, H.Y., Ling, Z.J., "Implementation of ITU-T G.729 speech codec
in IP telephony gateway", Wuhan University Journal of Natural Sciences, 2000, vol.5, no. 2, pp. 159-163.
[6] Feng, H.Y., Yang, J., Zhang, S., Zhang, J.L., "Implementation of ITU-T
G.723.1 dual rate speech codec based on TMS320C6201 DSP", In
Proceedings of 5th International Conference on Signal Processing, 2000,
vol. 2, pp.679-682.
[7] Han, S., Jeong, S., Yang, H., Kim,J., "Noise reduction for VoIP speech
codecs using modified Weiner filter", Advances and Innovations in
Systems, Computing Sciences and Software Engineering, 2007, pp. 393-
397
[8] Yamada, M. and Nishihara, A., "High-Speed FIR Digital Filter with
CSD Coefficients Implemented on FPGA", in Proceedings of IEEE
Design Automation Conference, 2001, pp. 7-8.
[9] Soderstrand, M.A., Johnson, L.G., Arichanthiran, H., Hoque, M. and
ELANGOVAN, R., "Reducing Hardware Requirement in FIR Filter Design", in Proceedings IEEE International Conference on Acoustics,
Speech, and Signal Processing 2000, Vol. 6, pp. 3275 - 3278
[10] Martinez-P. J., Valls, T. Sansaloni, Pascual, A.P. and Boemo, E.I.., "A
Comparison between Lattice, Cascade and Direct Form FIR Filter
Structures by using a FPGA Bit-Serial DA Implementation", in
Proceedings of IEEE International Conference on Electronics, Circuits
and Systems, 1999, Vol. 1,pp. 241 - 244.
[11] Proakis, J.D., Manolakis, D.G., Digital Signal Processing: Principles,
algorithm and applications, 3rd ed., 2000.
[12] Ifeachor, E.C., Jervis, B.W., Digital Signal Processing: A Practical
Approach, 2nd., Pearson Education, 2003.
[13] Li, C., "Design and realization of FIR digital filters based on
MATLAB", In Proceedings of International Conference on Anti-Counterfeiting Security and Identification in Communication, 2010, pp.
101-104.
[14] Mahdi, A.M., and Othman, M.B., "An algorithm proposed for FIR filter
coefficients representation", International Journal of Applied Mathematics and Computer Sciences, Vol.4, No.1, 2007, pp: 24-30
[15] Bolot, J.C., "End-to-end packet delay and loss behavior in the internet",
In Proceedings of ACM Symposium on Communications Architectures,
Protocols and Applications, 1993, pp.289-298
[16] Sannech, H., Le, N.T.L., "Speech property-based FEC for Internet
Telephony applications", In Proceedings of the SPIE/ACM/SIGMM
Multimedia Computing and Networking Conference, 2000, pp.38-51.
[17] Hohlfeld, O., Rudiger.G, Halblinger, G., "Packet loss in real-time services: Markovian Models generating QoE impairments", In
Proceedings of the 16th International workshop on quality of service,
2008, pp: 239-248.
[18] Jelassi, S., Youssef, H., Hoene, C., Pujolle, G., "Voicing aware
parametric speech quality models over VoIP networks", In Proceedings
of the Second international conference on Global Information Infrastructure Symposium, 2009, pp. 120-127.
[19] Wu, C.C., Chen, K.T., Huang, C.Y., Lei, C.L., “An Empirical evaluation
of VoIP playout buffer dimensioning in Skype, Google Talk and MSN
messenger”, In Proceeding of 19th International Workshop on Network
and Operating System Support for Digital Audio and Video, 2009, pp-
97-102.
[20] Singh, H.P., Singh, S., Singh, J., “Computer modeling and performance
analysis of VoIP under different strategic conditions”, in Proceedings of
2nd IEEE International Conference on Computer Engineering and
Applications, 2010, vol.1, pp. 611-615.
[21] Singh, H.P., Singh, S., Singh, J., “Digital signal processing approach for
performance improvement in Voice over Internet Protocol (VoIP)
network”, International Journal of Information and Communication
Technologies, 2011, vol.4, no. 1-2, pp. 85-90.
[22] http://www.utdallas.edu/~loizou/speech/noizeus/.
[23] 3GPP.TS 26.090 : Mandatory speech codec processing functions:
Adaptive Multi-Rate (AMR) speech codec: Transcoding functions, 2009
[24] ITU-T Recommendation G.729 Annex A. Reduced complexity 8 kbit/s
CS-ACELP speech codec, 1996.
[25] ITU-T Recommendation P.862: Perceptual evaluation of speech quality
(PESQ); an objective method for end-to-end speech quality assessment
of narrowband telephone networks and speech codec, 2001.
[1] Goralski, J.W., Kolon, C.M., IP Telephony, 1st ed., McGraw-Hill, 2000.
[2] Varshney, U., Snow, A., Mcgivern, M., Howard, C., "Voice over IP",
Communications of the ACM, 2002, vol.45,no.1, pp.89-96
[3] Verhelst, W., Roelands, M., "An overlap-add technique based on
waveform similarity (WSOLA) for high quality time-scale modification of speech", In Proceedings of IEEE International Conference on
Acoustics, Speech and Signal Processing, 1993, vol.2, pp. 554-557.
[4] Stenger, A., Younes, K.B., Girod, B., Sanneck, H., "A new technique for
audio packet loss concealment", In Proceedings of IEEE Global
Telecommunication Conference, Communications: The key to Global Prosperity, 1996, pp. 48-52.
[5] Feng, H.Y., Ling, Z.J., "Implementation of ITU-T G.729 speech codec
in IP telephony gateway", Wuhan University Journal of Natural Sciences, 2000, vol.5, no. 2, pp. 159-163.
[6] Feng, H.Y., Yang, J., Zhang, S., Zhang, J.L., "Implementation of ITU-T
G.723.1 dual rate speech codec based on TMS320C6201 DSP", In
Proceedings of 5th International Conference on Signal Processing, 2000,
vol. 2, pp.679-682.
[7] Han, S., Jeong, S., Yang, H., Kim,J., "Noise reduction for VoIP speech
codecs using modified Weiner filter", Advances and Innovations in
Systems, Computing Sciences and Software Engineering, 2007, pp. 393-
397
[8] Yamada, M. and Nishihara, A., "High-Speed FIR Digital Filter with
CSD Coefficients Implemented on FPGA", in Proceedings of IEEE
Design Automation Conference, 2001, pp. 7-8.
[9] Soderstrand, M.A., Johnson, L.G., Arichanthiran, H., Hoque, M. and
ELANGOVAN, R., "Reducing Hardware Requirement in FIR Filter Design", in Proceedings IEEE International Conference on Acoustics,
Speech, and Signal Processing 2000, Vol. 6, pp. 3275 - 3278
[10] Martinez-P. J., Valls, T. Sansaloni, Pascual, A.P. and Boemo, E.I.., "A
Comparison between Lattice, Cascade and Direct Form FIR Filter
Structures by using a FPGA Bit-Serial DA Implementation", in
Proceedings of IEEE International Conference on Electronics, Circuits
and Systems, 1999, Vol. 1,pp. 241 - 244.
[11] Proakis, J.D., Manolakis, D.G., Digital Signal Processing: Principles,
algorithm and applications, 3rd ed., 2000.
[12] Ifeachor, E.C., Jervis, B.W., Digital Signal Processing: A Practical
Approach, 2nd., Pearson Education, 2003.
[13] Li, C., "Design and realization of FIR digital filters based on
MATLAB", In Proceedings of International Conference on Anti-Counterfeiting Security and Identification in Communication, 2010, pp.
101-104.
[14] Mahdi, A.M., and Othman, M.B., "An algorithm proposed for FIR filter
coefficients representation", International Journal of Applied Mathematics and Computer Sciences, Vol.4, No.1, 2007, pp: 24-30
[15] Bolot, J.C., "End-to-end packet delay and loss behavior in the internet",
In Proceedings of ACM Symposium on Communications Architectures,
Protocols and Applications, 1993, pp.289-298
[16] Sannech, H., Le, N.T.L., "Speech property-based FEC for Internet
Telephony applications", In Proceedings of the SPIE/ACM/SIGMM
Multimedia Computing and Networking Conference, 2000, pp.38-51.
[17] Hohlfeld, O., Rudiger.G, Halblinger, G., "Packet loss in real-time services: Markovian Models generating QoE impairments", In
Proceedings of the 16th International workshop on quality of service,
2008, pp: 239-248.
[18] Jelassi, S., Youssef, H., Hoene, C., Pujolle, G., "Voicing aware
parametric speech quality models over VoIP networks", In Proceedings
of the Second international conference on Global Information Infrastructure Symposium, 2009, pp. 120-127.
[19] Wu, C.C., Chen, K.T., Huang, C.Y., Lei, C.L., “An Empirical evaluation
of VoIP playout buffer dimensioning in Skype, Google Talk and MSN
messenger”, In Proceeding of 19th International Workshop on Network
and Operating System Support for Digital Audio and Video, 2009, pp-
97-102.
[20] Singh, H.P., Singh, S., Singh, J., “Computer modeling and performance
analysis of VoIP under different strategic conditions”, in Proceedings of
2nd IEEE International Conference on Computer Engineering and
Applications, 2010, vol.1, pp. 611-615.
[21] Singh, H.P., Singh, S., Singh, J., “Digital signal processing approach for
performance improvement in Voice over Internet Protocol (VoIP)
network”, International Journal of Information and Communication
Technologies, 2011, vol.4, no. 1-2, pp. 85-90.
[22] http://www.utdallas.edu/~loizou/speech/noizeus/.
[23] 3GPP.TS 26.090 : Mandatory speech codec processing functions:
Adaptive Multi-Rate (AMR) speech codec: Transcoding functions, 2009
[24] ITU-T Recommendation G.729 Annex A. Reduced complexity 8 kbit/s
CS-ACELP speech codec, 1996.
[25] ITU-T Recommendation P.862: Perceptual evaluation of speech quality
(PESQ); an objective method for end-to-end speech quality assessment
of narrowband telephone networks and speech codec, 2001.
@article{"International Journal of Electrical, Electronic and Communication Sciences:55815", author = "H. P. Singh and S. Singh", title = "Computationally Efficient Signal Quality Improvement Method for VoIP System", abstract = "The voice signal in Voice over Internet protocol (VoIP) system is processed through the best effort policy based IP network, which leads to the network degradations including delay, packet loss jitter. The work in this paper presents the implementation of finite impulse response (FIR) filter for voice quality improvement in the VoIP system through distributed arithmetic (DA) algorithm. The VoIP simulations are conducted with AMR-NB 6.70 kbps and G.729a speech coders at different packet loss rates and the performance of the enhanced VoIP signal is evaluated using the perceptual evaluation of speech quality (PESQ) measurement for narrowband signal. The results show reduction in the computational complexity in the system and significant improvement in the quality of the VoIP voice signal.
", keywords = "VoIP, Signal Quality, Distributed Arithmetic, Packet Loss, Speech Coder.", volume = "6", number = "11", pages = "1303-6", }
