How to calculate sampling rate from frequency

The sampling rate is the number of times per second that an analog signal is sampled. The frequency is the number of times per second that a sound wave vibrates. The sampling rate must be at least twice the highest frequency of the analog signal in order to avoid aliasing. Aliasing is a type of distortion that can occur when a digital audio signal is sampled at a frequency that is too low.

To calculate the sampling rate from frequency, you can use the following formula:

Sampling Rate = Fs = 2fmax

where:

• Fs is the sampling rate in hertz (Hz)
• fmax is the highest frequency of the analog signal in hertz (Hz)

For example, if the highest frequency of the analog signal is 20 kHz, then the sampling rate must be at least 40 kHz. This is because the Nyquist Theorem states that the sampling rate must be at least twice the highest frequency in order to accurately represent the signal.

Here is an example of how to calculate the sampling rate from frequency:

def calculate_sampling_rate(fmax):
  """Calculates the sampling rate from frequency.

  Args:
    fmax: The highest frequency of the analog signal in hertz (Hz).

  Returns:
    The sampling rate in hertz (Hz).
  """

  sampling_rate = 2 * fmax
  return sampling_rate

if __name__ == "__main__":
  fmax = 20000
  sampling_rate = calculate_sampling_rate(fmax)
  print(f"The sampling rate is {sampling_rate} Hz.")

This code will calculate the sampling rate from frequency and print the result to the console.

Here are some additional things to keep in mind about calculating the sampling rate from frequency:

• The sampling rate must be an integer. This means that the sampling rate cannot be a decimal number.
• The sampling rate must be greater than or equal to twice the highest frequency of the analog signal.
• The sampling rate is typically specified in hertz (Hz). However, it can also be specified in other units, such as samples per second (sps).

I hope this helps!