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Sample Rate Conversion Algorithm

Sample rate conversion is algorithm of changing number samples (measurements) of signal level per second without changing waveform of the signal.

This kind of signal processing quality loss always.

In resulting signal appear non-linear distortions (noise burned by signal).

In high qualitative sample rate conversion algorithm the noise has level low -160 dB.

As comparing, studio audio cards has noise level about -120 dB.

Therefore noises of audio card masks conversion noise.


The algorithm steps are:

     1) Increase up of input sampling rate to frequency, multiple output signal's sampling frequency.

     2) Filtration of "parasitic" signals (non-linear distortions named as "artefacts") above half of target sample frequency.

     3) Decreasing sampling rate as multiple rejection of samples.

Sampling rate conversion algorithm details described here...

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How to Improve Sound Quality


In this article term "to improve sound quality" mean "to decrease sound playback distortions at analog output of digital-analog converter". How to try to decrease the playback distortions for used equipment or portable device (digital audio player, mobile phone, etc.) will be considered below.

AuI ConverteR 48x44 - Free Music Converter
AuI ConverteR 48x44 - Free Audio Converter

AuI ConverteR 48x44 is pro audio converter of music files DSF, DFF, ISO (1-bit audio), FLAC, WAV, AIFF, m4a, CD-audio, CUE, APE, ogg, mp3. The converter is intended for music production and audiophiles.

AuI ConverteR 48x44 FREE may be used as free audio converter for Windows and Mac OS X. There is professional digital audio processing (altering resolution - sample rate, bit depth and dithering) of audio files is provided.


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