Sample Rate Converter [AuI ConverteR 48x44 software]
audio file converter for music production and hi‑end audio

Sample Rate Converter [AuI ConverteR 48x44 software]

Audio signal consists of a sequence of samples. Sample is measurement of digital signal into point of time. They follow one to one with equal time interval.


What does a sample rate converter do?

Audio sample rate converter is intended for altering of time interval between signal samples. Signal is not changed. There are changed intervals between samples and its values.

Sample rate is digital-signal-sample number (measurements) per second. Higher sample rate is more samples per second.

Where sampling rate conversion is used        Sample rate conversion algorithm

Lo-end vs Hi-end converter     Resume

What is the use of sampling rate converter?

     Sample rate conversion uses in real time ("audio stream on fly") or during converting audio resolution of files.
     Real time change frequencies of digitization at playing of samples, and mix of several program-sequencer's audio tracks (imported from external files with different sampling frequencies).
     For audio equipment basically are 2 ranges of sampling frequencies:
    1) CD-audio: 44 100, 88 200, 176 400 Hz
    2) DVD-audio and DVD-video: 48 000, 96 000, 192 000 Hz
    To apply sampling rate conversion it is not necessary only for musicians and professional sound engineers, but also for home audio-video. For example, media player can adjust sample frequency (transparently for the user) for played audio files to frequency of the digitization in setting of sound card.


How to convert sample rate

To convert/change sample rate:

  1. Start sample rate converter software AuI ConverteR 48x44.

  2. Open an audio file file.
  3. At the main window, Format panel, choose output audio format.
  4. Set sample rate.
  5. Set bit depth value.
  6. If bit depth is set 16 bit, set dither ON.
    (for 24 bit and more resolutions, state of Dither button is ignored)
  7. Choose directory for output files (check out video, read guide)
  8. Click on the Start button.
  9. Wait until the end of conversion.

Remark: Free version has some limitations.




Watch and share: What inside HD audio converter?

How to work sample rate converter

      The algorithm of sampling frequency changing consists of following steps:
     1) Increase up of sampling rate (oversampling) to frequency, multiple target signal's sampling frequency.

     2) Filtration of "parasitic" signals (named as "artifacts") above half of target frequency of digitization.

     3) Make multiple rejection of superfluous samples for decreasing sampling rate (downsampling).


Sample Rate Conversion

      Increase of sample frequency is made by way of inserting additional ("virtual" - generated by interpolator) samples between "real" samples of input digital signal.


     The inserting of "virtual" samples, containing zero level values, is sometimes applied. It is faster method for calculations. But such way of increase of frequency of digitization adds a significant amount of "artifacts" to inevitable "artifacts" of precisely interpolated signal.

     For what it is necessary to increase frequency of digitization? For execute algorithm's item 3). Rejection of samples most easier it is multiple - simply delete superfluous.

     Further "parasitic" signals (with frequencies above half of target sampling frequency) are filtered. Otherwise (at deleting of "superfluous" samples) "artifacts" will mixed with spectrum of a useful signal and will distort it ("bad" sounds will added).



Does sample rate conversion affect sound quality?

     For reduction of distortions, added into converted signal, we should interpolate it as much as possible precisely. Accuracy of interpolation it is maximal degree of coincidence of initial analog signal with interpolator's "virtual" samples. It is necessary to remember, that the top-quality interpolator can restore an initial analog signal precisely enough. But not with 100 % accuracy. Alas. At increase of sampling frequency there will be "parasitic" signals above half of output signal's frequency of digitization.

     Special attention is paid to quality of low frequencies filter during development of hi-end sampling rate converters. If this filter will not suppress "artifacts", it will mixed with useful signal after deleting of "superfluous" samples.

Spectrum convergention

     For demonstration of filtration's quality we shall look at the time-diagram of spectrum. Horizontal axis there is time, vertical axis - frequency. Vertical lines - is spectrum into time point at horizontal axis. The level of signal is shown by color (white - the highest, black - the lowest - less minus 150 dB). Sinusoidal audio signal with sweep frequency (height of tone) come to input of sample rate converter.


Spectrum analyzis

     Here such result will be on an output of hi-end audio converter:

 Hi-end sampling rate conversion

     We see only recurrence of input signal, without in addition appearing frequency components - "artifacts". Level of artifacts less minus 150 dB. Less quantizations error level for 24-bit samples (minus 144 dB). Inevitable loose quantizations errors will mask "artifacts".

     The high and middle quality audio converter of sampling frequency will give a following picture:

Hi and middle quality sampling rate converter

     Here shown "artifacts" having a level near minus 105-110 dB (dark blue color) . These "artifacts" arise, both at interpolation, and at insufficient suppression (before decreasing sampling rate) of the "parasitic" signals located above half of frequency of digitization.

     Let's look for comparison the spectral diagram of too low quality converter:

 Lo-end sampling rate converter

     In this case "artifacts" achieve a level near minus 50 … 60 dB (red and pink colors).


Reasmpling filter

     The filter should pass audio a signal at different frequencies from 0 up to 20000 Hz without changing loudness useful. For this purpose inflating of the frequency characteristic (on different frequencies at passage through the filter) should not exceed change of a level of loudness of output signal less 1 … 2 dB.

Amplitude frequency response

     As much as possible to keep waveform of transformed audio signal it is required to provide an identical time delay for all spectral components at passage through the filter. It is provided, if the filter have linear phase-frequency characteristic. Linear is means in the form of a flat inclined line.

Phase frequancy characteristic filter low frequencies

     At such form of the phase characteristic all spectral components have an identical delay. The signal passes through the filter not deformed{distorted}. It is especially important for maintenance of quality sounds with sharp attack (drums, piano, guitar, etc.).

     Besides filters have such feature, as "ringing". It parasitic filter's output self-oscillation by sharply changing of input signal. Apply impulse to input of converter. At output the impulse turns to the oscillation stretched on time. "Ringing" is heard as click.

Resampling filgter ringing

     The more "sharp" recession to a level between pass bands and suppression band of filter, the more level of "ringing".
Therefore it is necessary for developer of the converter to choose the compromise between a sharpness of recession of filter's amplitude-frequency characteristic (that in the most positive influences to suppression of "artifacts") and level of "ringing". Very heavy case, when final result of conversion is sample rate 44,1 kHz. Between the maximal frequency of a useful signal (20 kHz) and half of frequency of digitization (22,05 kHz) the difference on frequency makes only 2,05 kHz. At a desirable degree of suppression of artifacts nearby 140 dB!!!



      Quality of sampling rate conversion depends at:
- quality of interpolation;
- value of oversampling frequency;
- quality of filtration before decreasing of sampling rate.

      For the majority of modern records 24-bit samples is used. It achieve theoretical level of quantizations noise minus 144 dB. Accordingly the level of all transformation's "artifacts" should not exceed minus 144 dB. Thus "artifacts" will masked in noise of quantizations. There is no reasons make level of "artifacts" less noise of quantizations.

     Quality of converting signal is provided with filter's linearity phase and evenness of frequency response.

See description high precision sample rate converter AuI ConverteR 48x44.



Frequently Asked Questions

How do you convert sample rate?

You can convert sample rate according to the manual...


Is 44.1 kHz a good sample rate?

Samplin rate 44.1 kHz is minimum discretization frequency to transmit audible range. Minimum frequency causes additional issues for developers of audio equipment and software.





Read about sampling rate



More information

August 6, 2022 updated | since July 3, 2011

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