PCM technique is used to convert analog voice signals into digital. In PCM the analog frequency is first sampled and then converted into binary bits. Each samples are taken as 8bits long. Basic communication theory requires that a minimum sampling rate of twice the frequency of the signal to be sampled will result in an accurate representation of the original signal.Human voice can have max 4000hz frequency, therefore sampling rate should be 8000 samples/sec.Which implies required bit rate for transmitting voice is 8000*8 = 64000 bits/sec = 64kbps.
to shift the frequency of information signal ,at the frequency domain to a higher frequency ...so the information can be transmitted to the receiver.
repetition rate of signal
The frequency domain of a voice signal is normally continuous because voice is a nonperiodic signal.
A low pass signal whose bandwidth is much smaller than its center frequency, such as an AM signal. It is a a signal with its spectrum concentrated around zero frequency.
The Nyquist frequency should not be confused with the Nyquist rate, which is the minimum sampling rate that satisfies the Nyquist sampling criterionfor a given signal or family of signals. The Nyquist rate is twice the maximum component frequency of the function being sampled. For example, the Nyquist rate for the sinusoid at 0.6 fs is 1.2 fs, which means that at the fs rate, it is being undersampled. Thus, Nyquist rate is a property of a continuous-time signal, whereas Nyquist frequency is a property of a discrete-time system.When the function domain is time, sample rates are usually expressed in samples/second, and the unit of Nyquist frequency is cycles/second (hertz). When the function domain is distance, as in an image sampling system, the sample rate might be dots per inch and the corresponding Nyquist frequency would be in cycles/inch.
A 20Hz signal must be sampled at a minimum of 40Hz to have a chance of sampling both peaks and to get a reasonable representation it must be sampled at a minimum of 100Hz.For a sampling rate of 30Hz the Nyquist frequency is 15Hz and since 20Hz is above that it will generate the alias signal of 10Hz in the sampled data instead of the original signal of 20Hz. Therefore it is not possible to do what you ask.
If you sample at more than the Nyquist frequency (one half the signal frequency) you introduce an aliasing distortion, seen as sub harmonics.
The frequency range for a voice signal is 300 to 4000 HZ. The Nyquist theorem states a waveform should be sampled at 2 times its highest frequency. So 2 x 4000 is 8000 samples every second. This ensures an adequate representation of the signal.
if the sampling rate is twice that of maximum frequency component in the message signal it is known as nyquist rate
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.
Nyquist theorem is applicable to both optical fiber and copper wire systems. It states that the sampling rate must be at least twice the highest frequency component of the signal being sampled to accurately reconstruct the signal. In optical fiber systems, this principle is key to ensuring the integrity and accuracy of data transmission.
The Nyquist frequency for a signal with a maximum bandwidth of 1 KHz is 500 Hz, however that will lead to aliasing unless perfect filters are available. The Nyquist rate for a signal with a maximum bandwidth of 1 KHz is 2 KHz, so the answer to the question is 2 KHz, or 500 microseconds.
It states that for satisfactory representation of the sampled signal the sampling frequency must be atleast equal to twice the highest input freq, which is called nyquist sampling. If its less than twice, undersamplin occurs resulting in distortion.
Sampling a signal is a process where some thing, usually an analog signal, is sampled at a particular frequency, and analysis and processing is performed on that sample stream. The advantages are that conversion between the time domain and the frequency domain using Fourier analysis is a very powerful technique that allows you to do a lot of different things, such as compression and filtering, without needing advanced electronics. You can also transmit the digital samples from one point to another, using digital electronics, rather than analog electronics. Disadvantages are that sampling, by its very nature, introduces distortion, because you have limited resolution on the ADC and you have limited frequency of sample rate. Often, however, you can do very well, so long as you understand the implications of sampling. One of them is called Nyquist Aliasing. That is where the sample rate is less than the Nyquist frequency of one half of the highest harmonic of the signal. This is a very noticeable distortion, perceived as a buzzing in inverse frequency terms, which must be properly filtered. In fact, if you look at a traditional audio CD, the sample rate is 44.1 KHz, making the Nyquist frequency 22.05 Khz. That is commonly above the range of human hearing, but you still must account for it, otherwise, the distortion could damage the equipment or degrade the signal quality.
The 44.1KHz frequency is the frquency of the 'sampling rate' With digital audio the original analog signal has to be 'sampled' ie measured. These samples measure the amplitude (strength) of the signal at that particular point in time. Since the analogue signal is constantly changing the sampling must be done continuously. The sampling rate must ALWAYS be higher than the highest frequency of the source signal, (otherwise it cant be measured). There was a bloke called Nyquist who postulated that the sample rate must be at minimum twice the highest frequency of the sample. This has been accepted as common practice now. This is where we get 44K (ie 44,000 measurements or samples per second), which is about double the 20K audio source.
The Nyquist Therorem states that the lowest sampling rate has to be equil to or greather than 2 times the highest frequency. Therefore the sampling rate should be 400Hz or more.