Prices. Choose optimal >
Order >
Advantages
F.A.Q.
Warranty
Support
Contact

Audio Sample Rate Conversion

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.

Sample rate conversion (SRC) audio is changing time interval between samples of signal. Signal not changed. Changed intervals and values of samples.

Sample rate is count per second of samples (measurements) of digital signal. More sample rate - more samples per second.


sample rate conversion audio

Where sampling frequency converting is used


     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.

 

What inside HD audio converter?

Sample rate conversion algorithm

      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 "artefacts") 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.

Interpolation

     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 "artefacts" to inevitable "artefacts" 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) "artefacts" will mixed with spectrum of a useful signal and will distort it ("bad" sounds will added).

Hi-end sampling rate audio converter
vs
middle-quality converter

     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 "artefacts", it will mixed with useful signal after deleting of "superfluous" samples.

Convergention spectrum

     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.

     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 - "artefacts". Level of artefacts less minus 150 dB. Less quantizations error level for 24-bit samples (minus 144 dB). Inevitable loose quantizations errors will mask "artefacts".

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

Hi and middle quality sampling rate converter

     Here shown "artefacts" having a level near minus 105-110 dB (dark blue color) . These "artefacts" 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 "artefacts" achieve a level near minus 50 … 60 dB (red and pink colors).

     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 frequancy characteristic

     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.

Ringing of filter

     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 "artefacts") 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 artefacts nearby 140 dB!!!

Resume

      Quality of sampling rate conversion depend 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 "artefacts" should not exceed minus 144 dB. Thus "artefacts" will masked in noise of quantizations. There is no reasons make level of "artefacts" 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.

 

Read also:

About reasons of using optimal sample rate (pdf).

[Login/Register]
AuI ConverteR 48x44 RSS Audiophile Inventory on Facebook Audiophile Inventory on Google+ Audiophile Inventory on LinkedIn

Audiophile converter
ISO DSF DFF FLAC WAV...
 CD ripper
DSF/FLAC metatag editor
AuI ConverteR 48x44

 
Loading

Released new studio audio converter AuI ConverteR 48x44 v. 6.1 CD ripper, ISO, DFF, DSF, WAV, FLAC, AIFF, ALAC. Apllied input and ouput DSF (D1024) audio file support; non-English simbols in ISO file name issues (under Windows), other minor issues.

Released new studio audio converter AuI ConverteR 48x44 v. 6.0.3 CD ripper, ISO, DFF, DSF, WAV, FLAC, AIFF, ALAC. In the settings added option for extracting of stereo or multichannel track only. Fixed wrong reading of long album, performer, track names from CD database.

Released new studio audio converter AuI ConverteR 48x44 v. 6.0.2 CD ripper, ISO, DFF, DSF, WAV, FLAC, AIFF, ALAC. Added album artwork management, fixed CUE+multiple files issue, trimmed long file path/name for ISO extractor under Windows and several minor bugs.

Copyright © Yuri Korzunov [Audiophile Inventory (audiventory.com)], 2010-2016. All Rights Reserved.

Site map