Rate refers to frequency, while size refers to the amount.
Thus, Sampling Rate is measured in Hertz (number of times per second a sample is taken), and Sampling Size is measured in Bits (number of binary digits of information taken at a single time).
Thus, if you Sample at 10 Hz/8 bits, that means you take 8 bits of information, 10 times per second.
The ADC10 peripheral on most MSP430s can sample at upwards of 200 ks/s.
Oversampling is part of signal processing. It is the process of using a sampling frequency that is higher than the Nyquist rate to sample a signal.
The data rate for Pulse Code Modulation (PCM) digital voice signals is calculated using the formula: Data Rate = Sampling Rate × Bits per Sample. For standard voice applications, the sampling rate is typically 8 kHz (samples per second), and each sample is represented by 8 bits (1 byte). Thus, the calculation is 8,000 samples/second × 8 bits/sample = 64,000 bits per second, or 64 Kbps. This rate is sufficient to capture the frequency range of human speech effectively.
Sampling rate or sampling frequency defines the number of samples per second (or per other unit) taken from a continuous signal to make a discrete or digital signal.
The sampling rate must be at least double the highest frequency component of the modulating signal in order to avoid frequency aliasing.
frequency is simply the rate at which something is happening, ie the frequency of Christmas is once a year, the frequency of having breakfast is once a day etc. If frequency is expressed in Hertz, it's how many times something happens during a second. Sampling is, well, sampling. Usually means testing and measuring something changeable. If you're running a bath and occasionally stick your fingers in to check the temperature, then that's sampling, Sampling frequency simply describes at which rate you're making whatever test or measurement it is you're talking about.
You must sample at 2 x the rate of the analog signal (2 x the analog signal frequency).
The ADC10 peripheral on most MSP430s can sample at upwards of 200 ks/s.
sample rate difference. they have to be set to the same sample rate
A sampling rate is a term that is used in digital recording to describe how much, and how often, data is used. In digital audio (sound recording), a new sample of analog data -- a new speaker position -- is sent out to the speaker quite often, usually at a sample rate of 44100 Samples/second. So when the music is recorded for a CD, a new sample is collected from the microphones just as often, usually at a sample rate of 44,100 Samples/second. A biologist may measure the temperature of a lake once a week. That temperature data has a sampling rate is 1 Sample/week. Sampling rate is independent of "channels" or "bit resolution". A highly instrumented concert may have a dozen channels, each one from a microphone sampled at 44100 Samples/second, but the total sampling rate is still 44,100 Samples/second.
not sure what your asking, but if you are asking what i think your asking, you have to sample at least at twice bandwidth of the frequency you are sampling. This is known as Nyquist Rate http://en.wikipedia.org/wiki/Nyquist_rate
Decimation in digital processing is the process wherein sampling rate of a signal is reduced. An increase in sample rate is complementary to interpolation.
The sampling rate is the number of samples taken from a continuous signal over a period of time (typically measured per second - Sa/s or Samples per Second). The Nyquist - Shannon Sample Theorem states that a sample rate should be double the highest recorded frequency. Since the range of human hearing is 20Hz - 20,000 Hz, the minimum sample rate should be 40,000 Hz. CD format sample rates are 44.1kHz for this reason as well as other technical reasons.
I would like to sample the signal Xa(t) =1+cos(10 *pi*t) using sampling frequency fs=8 Hz. How can I calculate this? ANSWER: Your signal has a frequency component of 5hz (from the equation: 2*pi*f*t = 10*pi*t, therefore f=5). The Nyquist rate for this signal (the minimum sampling rate required to reconstruct the signal) is then 10Hz, and even at that rate the amplitude of the sampled signal will be reduced unless you can somehow synchronize the sampling with the peaks/troughs of the cosine signal. If you sample at 8Hz you will not be able to reconstruct the signal at all.
Active Sampling refers to the process of connecting sample collection media (i.e., thermal desorption tube) to a sampling train consisting of inert tubing and a sample pump operating at a know flow rate. Once activated, the sample pump draws air through the sample media, resulting in a know sample volume. Passive Sampling refers to the process of allowing sample collection media (i.e., thermal desorption tube) to passively diffuse air through the sample media without benefit of 'forced air'. This allows for longer potential sample periods without concern for overloading the sampling media. •
It signals the difference between successive sample sizes
The minimum sample rate required to record a frequency of 96 kHz is 192 kHz. This is because according to the Nyquist theorem, the minimum sampling rate must be at least twice the highest frequency in order to accurately reconstruct the original signal. So for a frequency of 96 kHz, the minimum required sampling rate is double, which equals 192 kHz.