A: It reference to processes of data gathering from a signal. Like telemetry of heat, and other parameters that the signal may represent,
sampling is a one type of process use for converting into analog signal to digital signal.
Upsampling is the process of increasing the sampling rate of a signal. For instance, upsampling raster images such as photographs means increasing the resolution of the image.In signal processing, downsampling (or "subsampling") is the process of reducing the sampling rate of a signal. This is usually done to reduce the data rate or the size of the data.
Sampling Theorum is related to signal processing and telecommunications. Sampling is the process of converting a signal into a numeric sequence. The sampling theorum gives you a rule using DT signals to transmit or receive information accurately.
Sampling rate is a defining characterstic of any digital signal. In other words, it refers to how frequently the analog signal is measured during the sampling process. Compact disks are recorded at a sampling rate of 44.1 kHz.
while conversion of analog signal to digital signal, we need to convert continuous analog signal to discrete signal. this can be done by dividing the analog signal into specific time slots. this process is known as sampling. there is a condition for sampling that can be given as follows. fs<=2fm
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.
Instantaneous sampling is one method used for sampling a continuous time signal into discrete time signal. This method is called as ideal or impulse sampling. In this method, we multiply a impulse function with the continuous time signal to be sampled. The output is instantaneously sampled signal.
sampling theorem is used to know about sample signal.
In signal processing, sampling is the reduction of a continuous signal to a discrete signal. A common example is the conversion of a sound wave (a continuous signal) to a sequence of samples (a discrete-time signal).
state and prove sampling theory as applied to low pass signal
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.
DSP -Digital signal processing the word it self says it is process of analyzing the digital signal . The system which does this work of analysis is the digital signal processing system. In this system the raw analog signal(ex:human voice) which is time domain signal is fed to the pre aliasing filter which removes noise part of the signal and this fed to the sampling and quantization in which the real time analog signal is discretized and digitized . After this process this digitized signal is fed to the DSP processor to do the specific operation on the signal (ex: convolution , modulation etc.,) .This processed signal is fed to the D/A which converts the digital signal to the analog signal real time signal. after all explaining the process of signal processing. the stages between the sampling and D/A is the digital processing system.