Increasing the Accuracy of Signal Formation by Changing the Sampling Rate

Abstract

The problem of changes in the audio signal when changing the sampling frequency arose a long time ago, but interest in it has recently become more acute with an increase in the quality of signal formation and transmission. To some extent, this was also determined by the methods of assessing the quality of changes in the harmonic signal without taking into account changes in the actual broadcast signal. Algorithms for the formation of a new digital sequence after changing the sampling rate provided perfect quality in the conversion of the sinusoidal signal and large errors in the conversion of the real signal. Algorithms for changing the sampling rate Fd have been improved. The aim of this work is to study the features of changes in the sampling frequency in the paths of the broadcast channel and to develop algorithms for converting the sampling frequency in the frequency domain with an increase in conversion accuracy. It is shown that the algorithms for converting the sampling rate in the time domain have reached perfection and cannot be improved, so the transition to processing in the frequency domain is important. An algorithm for converting the sampling frequency in the frequency domain has been developed. The spectrum of the error signal for various types of converters is determined and the need for its compensation is determined. A method for introducing predistortions for converters with known and unknown sequence of sample rates used has been developed. The use of the results of this work makes it possible to improve the quality of broadcasting signals and information programs.

Keywords

• sound broadcast signal
• discrete fourier transform
• sample rate
• window function
• error signal spectrum
• oversampling distortion