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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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AuI ConverteR 48x44


Released new studio audio converter AuI ConverteR 48x44 v. 6.2 CD ripper, ISO, DFF, DSF, WAV, FLAC, AIFF, ALAC. Apllied downmix multichannel to stereo (5.1 to 2.0) and translation of user interface to Russian and German (partitially yet).

Final of the article about click matter into DSF and PCM (FLAC, WAV, AIFF, etc.) files after ISO (1-bit audio) extraction. Clicks inside audio stuff.

Available for testing new studio audio converter AuI ConverteR 48x44 v. 6.1.7-beta CD ripper, ISO, DFF, DSF, WAV, FLAC, AIFF, ALAC with downmix 5.1 to 2.0 (multichannel to stereo) ability.

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