Communications - Scientific Letters of the University of Zilina 2004, 6(4):110-112 | DOI: 10.26552/com.C.2004.4.110-112

Speech compression algorithm based on non-equidistant sampling

Igor Mihalik1
1 Department of Information Networks, FRI, University of Zilina, Slovakia

The method presented here is one of the methods of time domain compression. The technique uses non-equidistant sampling. Human voice and voice conversation are characterized by transitions. During the speech frequency charactericstics change so that viariable sampling can be applied. The nonuniform sampling method which is implemented is described in this paper; its usage in speech synthesis software which is being developed at the Department of Info-Com Networks is shown.

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Published: December 31, 2004  Show citation

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Mihalik, I. (2004). Speech compression algorithm based on non-equidistant sampling. Communications - Scientific Letters of the University of Zilina, 6(4), 110-112. doi: 10.26552/com.C.2004.4.110-112
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    References

    1. MINOLI, D., MINOLI, E.: Delivering Voice over IP networks, ISBN 0-471-25482-7, (1998)
    2. PULSE CODE MODULATION (PCM) OF VOICE FREQUENCIES ITU-T Recommendation G.711
    3. 40, 32, 24, 16 kbit/s ADAPTIVE DIFFERENTIAL PULSE CODE MODULATION (ADPCM) ITU-T Recommendation G.726
    4. 7 kHz AUDIO - CODING WITHIN 64 KBIT/S ITU-T Recommendation G.722
    5. Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3kBit/s ITU-T Recommendation G.723.1
    6. Engineering Fundamentals, Sampling Theorem and Nyquist Rate, http://www.efunda.com/designstandards/sensors/methods/DSP\_nyquist.cfm
    7. Black, A. W., Taylor, P., Caley, R.: "Festival Speech Synthesis System", http://www.cstr.ed.ac.uk/projects/festival/festival-toc.html
    8. Igor Mihálik: Návrh parametrov kompresného algoritmu, DP reg.číslo 275/2000, (2001).

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