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Determining the actual sampling frequency of your sound-card or chip-set

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  • Determining the actual sampling frequency of your sound-card or chip-set

    Many sound-cards or A/D conversion chip-sets claim sampling rates above 48 kHz. Often, the claim is made that they support up to 192 kHz. While this may be true (support) the system may or may not actually be sampling at rates higher than 48 kHz. Instead, the system emulates rates above 48 kHz. So, how do you know what your system is doing? Here is a method using your Diamond Cut Productions software to measure the sample rate based on what Nyquist teaches us about data sampling. We learn from his theorem that the sampling frequency of a system must be at least twice the maximum frequency to be accurately digitized. So, for example, a 44.1 kHz sampling system will yield under ideal conditions (perfect brick wall filters) an upper frequency limit of 22.05 kHz. The following uses the recording function of Diamond Cut Productions software along with it's spectrum analyzer. So, what do I do?

    1. Set your sound card up for a Microphone Input. This should not be set for the internal microphone, but the external one available via the external input jack on your system (often consisting of an 1/8 inch TRS {tip, ring sleeve} connector).

    2. Plug a cable into the mic input. Make sure that it is connected to nothing on the other end and make sure that the ends are not touching anything metal.

    3. Set up the Diamond Cut software recorder to the following parameters:
    A. Stereo
    B. 16 Bit Resolution
    C. 192 kHz Sampling Rate

    4. Now, record (silence) for ~20 seconds and save the file marking it for ease of retrieval.

    5. Bring up the file in your Diamond Cut Productions Software.

    5. Under the View menu, bring up the Spectrum Analyzer.

    6. Set up the Spectrum Analyzer as follows:
    A. FFT Size: 8192
    B. Frequency Resolution: 23.44 Hz
    C. Window: Bessel
    D. Amplitude Range: 50 dB
    E. Display Mode: Slow

    7. Using your mouse, highlight 2/3rds of the file leaving out the start and end of the recording (the signal will not be visible as the noise signal is buried in the mud - - - but it is there).

    8. Click on the "Loop Play" icon on the toolbar.

    9. A signal will start to build up on the spectrum analyzer. Use the Offset control to place the obvious 'step' in the middle of the screen.

    10. After the signal has built up for a few seconds, you can then click on the sharp drop section of the displayed signal using your left mouse button. The sharp drop-off is the action of the anti-aliasing filter in the sound-card or A/D chip-set.

    11. Look in the upper left corner of the Spectrum Analyzer Display and you will see a frequency value displayed.

    12. Your sound card sampling rate is approximately twice that value.

    13. For example, on this computer, it reads 23.5 kHz (despite the fact that I recorded at 192 kHz). Thus, the true sampling rate for my computer is 23.5 kHz x 2 = 47 kHz.

    14. While my computer system is supporting the 192 kHz sampling rate, it is emulating anything above around 47 kHz.

    15. I have seen A/Ds (sound-cards and/or chip-sets) that will truly sample at 205 kHz which yields a usable upper frequency limit of 100 kHz. These are used mostly in scientific applications and research.

    Note: If you are not confident that you are using a 192 kHz sampled file, goto the View Menu and click on "File Info" and you shall see what the file sample rate report is.

    Craig
    Last edited by Craig Maier; 03-07-2018, 11:04 PM.
    "Who put orange juice in my orange juice?" - - - William Claude Dukenfield

  • #2
    Hi...in my case I have set up my control loop to work in real-time at 2.5kHz to control a 1kHz tone. This sampling rate is defined by setting the scan rate of the input card (PXI-6052E)to 2500 scans/sec.
    The 1000Hz signal that is then acquired by the system also appears to have an amplitude modulation of a much lower frequency than than of the 1000Hz. The only reason that I can think of for this is that the analog input is actually being sampled a rate which is very close but slightly higher than 2.5kHz by a few samples per second.

    printed circuit assembly
    Last edited by NealXu; 02-05-2018, 03:17 PM.

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    • #3
      That could be a form of Intermodulation Distortion (low frequency AM) or some aliasing product.
      "Who put orange juice in my orange juice?" - - - William Claude Dukenfield

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      • #4
        It could be that the anti-aliasing filter in your sound card is fixed at 22 kHz. Ideally, it would be variable and set internally a little higher than twice the sampling frequency. In you case, you would want the filter (brick wall low-pass filter inside the sound-card) to be set at around 2.6 kHz.

        Craig
        Last edited by Craig Maier; 01-19-2018, 10:56 PM.
        "Who put orange juice in my orange juice?" - - - William Claude Dukenfield

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