In PCM an anlog signal or information is converted into a binary sequence, i. Basic Elements of PCM. The transmitter section of a Pulse Code Modulator circuit consists of Sampling, Quantizing and Encoding, which are performed in the analog-to-digital converter section.
The low pass filter prior to sampling prevents aliasing of the message signal. In Radio Broadcasting, a relatively large signal to noise ratio typically of the order of 60 dB is required.
Therefore, higher bandwidth requirement and complicated circuitry are the drawbacks of PCM which does not allow its use fr the radio, TV broadcasting applications. A digital technique that involves sampling an analog signal at regular intervals and coding the measured amplitude into a series of binary values, which are transmitted by modulation of a pulsed, or intermittent, carrier. Block diagram of Pulse Code Modulation. The figure below shows the block diagram representing a PCM system.
It is basically composed of a transmitter, a transmission path and a receiver. The transmitter performs the sampling, quantizing and encoding of the signal. The signals within 20 Hz to 20 kHz frequency range can travel only a few distances. To send the message signal, the length of the antenna should be a quarter wavelength of the used frequency.
Thus, modulation is required to increase the frequency of the message signal and to enhance its strength to reach the receiver. There is a uniform format used for the transmission of different types of basic band signals.
Hence is easy to integrate all the signals together and send them on the common network. Get Started for Free Download App. More Pulse Modulation Questions Q1.
Which of the following Pulse time Modulation does not exist in practice? What does DM stands for? Study the given input signal and match the columns. PAM 2. PWM 3. When analog signals become weak because of transmission loss, it is hard to separate the complex analog structure from the structure of random transmission noise.
If you amplify analog signals, it also amplifies noise, and eventually analog connections become too noisy to use. Digital signals, having only "one-bit" and "zero-bit" states, are more easily separated from noise. They can be amplified without corruption. Digital coding is more immune to noise corruption on long-distance connections.
Also, the world's communication systems have converted to a digital transmission format called pulse code modulation PCM. PCM is a type of coding that is called "waveform" coding because it creates a coded form of the original voice waveform. This document describes at a high level the conversion process of analog voice signals to digital signals. For more information on document conventions, refer to the Cisco Technical Tips Conventions.
The first step to convert the signal from analog to digital is to filter out the higher frequency component of the signal. This make things easier downstream to convert this signal. Most of the energy of spoken language is somewhere between or hertz and about or hertz. Roughly hertz bandwidth for standard speech and standard voice communication is established. Therefore, they do not have to have precise filters it is very expensive.
A bandwidth of hertz is made from an equipment point if view. This band-limiting filter is used to prevent aliasing antialiasing. The sampling frequency is less than the highest frequency of the input analog signal. This creates an overlap between the frequency spectrum of the samples and the input analog signal. The low-pass output filter, used to reconstruct the original input signal, is not smart enough to detect this overlap.
Therefore, it creates a new signal that does not originate from the source. This creation of a false signal when sampling is called aliasing.
The second step to convert an analog voice signal to a digital voice signal is to sample the Filtered input signal at a constant sampling frequency. It is accomplished by using a process called pulse amplitude modulation PAM. This step uses the original analog signal to modulate the amplitude of a pulse train that has a constant amplitude and frequency. See Figure 2. The pulse train moves at a constant frequency, called the sampling frequency. The analog voice signal can be sampled at a million times per second or at two to three times per second.
With AM radio, the amplitude, or overall strength, of the signal is varied to incorporate the sound information. With FM, the frequency the number of times each second that the current changes direction of the carrier signal is varied.
The signals within 20 Hz to 20 kHz frequency range can travel only a few distances. To send the message signal, the length of the antenna should be a quarter wavelength of the used frequency. Thus, modulation is required to increase the frequency of the message signal and to enhance its strength to reach the receiver. While QPSK is used for data transmission to provide higher data rate. Your email address will not be published. Save my name, email, and website in this browser for the next time I comment.
Skip to content How does pulse code modulation work? Why do we need Pulse Code Modulation? What do you mean by pulse modulation? Which modulation technique is most affected by noise? Which is a quantization process? Why is PCM more immune to noise? What are the disadvantages of PCM? Which is better PCM or Bitstream? Why companding is used in PCM? What is PCM and its advantages?
0コメント