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Real-Time Partitioned Convolution for Ambiophonics Surround Sound

By Anders Torger, University of Parma Industrial Engineering Dept. V.Scienze 181/A, 43100 PARMA, ITALY
Email: torger@ludd.luth.se

and Angelo Farina University of Parma Industrial Engineering Dept. V.Scienze 181/A, 43100 PARMA, ITALY
Email: farina@pcfarina.eng.unipr.it

4. Conclusions

The partitioned convolution algorithm presented here makes it possible to employ a standard, low-cost personal computer for implementing a complete Ambiophonics surround sound processor. This gives access to the Ambiophonics technology for a wide number of users. The same processing can also be used for other audio applications which require multi-channel convolution with very long impulse responses, such as multi-channel reverberation, wave front synthesis and detailed equalization of loudspeaker arrays.

The algorithm proved to be very efficient in terms of computer resources. In the currently available public domain implementation, it can provide higher throughput than the traditional unpartitioned overlap-and-save and the non-uniform partitioned convolution algorithms, which both are theoretically cheaper.

5. Acknowledgements

This work was supported economically and technically by the Ambiophonics Institute, founded by Ralph Glasgal. The authors express here their gratitude to and admiration for him, for having developed the Ambiophonics technology and having promoted it for mass deployment completely royalty-free and without patents, giving substantial support to the research on surround sound reproduction and advanced digital signal processing.

6. References

[1] A. Farina, R. Glasgal, E. Armelloni, A. Torger, “Ambiophonic Principles for the Recording and Reproduction of Surround Sound for Music”, 19th AES Conference, Schloss Elmau, Germany, 21-24 June 2001.

[2] R. Glasgal, K. Yates, “Ambiophonics – Beyond Surround Sound to Virtual Sonic Reality”, Ambiophonics Institute, 1995.

[3] O. Kirkeby, P. A. Nelson, H. Hamada, “The Stereo Dipole – A Virtual Source Imaging System Using Two Closely Spaced Loudspeakers”, J. AES vol. 46, n. 5, 1998 May, pp. 387-395.

[4] O. Kirkeby, P. Rubak, A. Farina, “Analysis of ill-conditioning of multi-channel deconvolution problems”, 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, Mohonk Mountain House New Paltz, New York October 17-20, 1999.

[5] J. J. Lopez, A. Gonzalez, “PC Based Real-Time Multichannel Convolver for Ambiophonic Reproduction”, 19th AES Conference on Surround Sound, Schloss Elmau, Germany, 21-24 June 2001.

[6] A. Torger, “BruteFIR – an open-source general-purpose audio convolver”, http://www.ludd.luth.se/˜torger/brutefir.html

[7] A. V. Oppheneim, R. Schafer, “Digital Signal Processing”, Prentice Hall, Englewood Cliffs, NJ 1975, p. 242.

[8] W. G. Gardner, “Efficient convolution without input-output delay”, J.AES vol. 43, n. 3, 1995 March, pp. 127-136.

[9] T. G. Stockham Jr., “High-speed convolution and correlation”, AFIPS Proc. 1966 Spring Joint Computer Conf., Vol 28, Spartan Books, 1966, pp. 229 - 233.

[10] J. S. Soo, K. K. Pang, “Multidelay block frequency adaptive filter”, IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-38, No. 2, February 1990.

[11] M. Frigo, S. G. Johnson, “FFTW: An Adaptive Software Architecture for the FFT”, Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, vol. 3, 1998, pp. 1381-1384.