Pulse Code Modulation

The modulation methods PAM, PWM, and PPM discussed in the previous lecture still represent analog communication signals since the height, width, and position of the PAM, PWM, and PPM, respectively, can take any value in a range of values. Digital communication systems require the transmission of a digital for of the samples of the information signal. Therefore, a device that converts the analog samples of the message signal to digital form would be required. Analog to Digital Converters (ADC) are such devices. ADCs sample the input signal and then apply a process called quantization. The quantized forms of the samples are then converted to binary digits and are outputted in the form of 1's and 0's. The sequence of 1's and 0's outputted by the ADC is called a PCM signal (Pulses have been coded to 1's and 0's). Example: A color scanner is scanning a picture of height 11 inches and width 8.5 inches (Letter size paper). The resolution of the scanner is 600 dots per inch (dpi) in each dimension and the picture will be quantized using 256 levels per each color. Find the time it would require to transmit this picture using a modem of speed 56 k bits per second (kbps). We need to find the total number of bits that will represent the picture. We know that 256 quantization levels require 8 bits to represent each quantization level. Number of bits = 11 inches (height) * 8.5 inches (width) * 600 dots / inch (height) * 600 dots / inch (width) * 3 colors (red, green, blue) * 8 bits / color = 807,840,000 bits Using a 56 kbps modem would require 807,840,000 / 56,000 = 14426 seconds of transmission time = 4 hours. For this reason, compression techniques are generally used to store and transmit pictures over slow transmission channels. Quantization The process of quantizing a signal is the first part of converting an sequence of analog samples to a PCM code. In quantization, an analog sample with an amplitude that may take value in a specific range is converted to a digital sample with an amplitude that takes one of a specific pre–defined set of quantization values. This is performed by dividing the range of possible values of the analog samples into L different levels, and assigning the center value of each level to any sample that falls in that quantization interval. The problem with this process is that it approximates the value of an analog sample with the nearest of the quantization values. So, for almost all samples, the quantized samples will differ from the original samples by a small amount. This amount is called the quantization error. To get some idea on the effect of this quantization error, quantizing audio signals results in a hissing noise similar to what you would hear when play a random signal. Assume that a signal with power P s is to be quantized using a quantizer with L = 2 n levels ranging in voltage from –m p to m p as shown in the figure below.

British Journal of Computer, Networking and Information Technology, 2021

The quality of the signal is essential in digital communication and signal processing. The transmission channel is also important. Modulation is used for effective transmission of signal. There exist several types of modulation techniques. One of such is the pulse code modulation (PCM). The performance of PCM is however affected by quantization error and noise in the transmission channel, which affects the quality of the output. Against this backdrop, this paper presents the use of differential pulse code modulation (DPCM) so as to address the limitation of pulse code modulation. The simulation environment is MATLAB 2018a. The MATLAB Simulink is used to design the PCM and DPCM systems using appropriate digital processing blocks. The DPCM system shows a significant improvement in terms of error reduction and quality of output.

Asia-pacific Journal of Multimedia Services Convergent with Art, Humanities, and Sociology, 2012

Voice-over Internet Protocol (VoIP) telephony is increasingly becoming a variant telecommunication technology that one day may surpass the old analog and digital telephone systems. The Quality-of-Service (QoS) factor is an important parameter to be considered when measuring the performance of a VoIP system. Many factors may influence the QoS of a given VoIP system. Some of these factors are delay, jitter and packet loss. This paper served to demonstrate that the algorithmic delay (latency) of the Pulse Code Modulation (PCM) speech compression algorithm which has originally been implemented in software can be significantly reduced by implementation in hardware. The PCM speech compression algorithm is first implemented, verified to demonstrate equivalence and then validated by comparing the latency of the hardware-implemented speech compression algorithm with that of an existing software implementation.

IJCSMC, 2019

The digital revolution has emerged the need for storing, processing and communicating data in digital format. Accordingly, all-digital systems have become popular, where no analogue components are included in the circuitry. As the output of these systems is digital, pulse code modulation is employed to establish communications without the need for analogue components. A certain characteristic of the pulse, such as its width, position or amplitude, is adjusted based on the input value. The number of bits transmitted in the channel depends on the number of bits modulated per pulse and the number of pulses outputted from the system per second. However, these parameters are limited by the capabilities of the digital system and communication channel. In this study, a new hybrid communications technique is proposed to allow increasing the bandwidth of digital communication systems, without the need for significant modification to the system. The bits of the input data is split into two sets, one modulated using Pulse Width Modulation and the other is modulated using Pulse Position Modulation. The outputs from these modulators are encoded to the output signal, using one of two possible encoding techniques, amplitude or frequency, depending on the characteristics of the communications channel. The evaluation results show that the proposed method has been able to provide twice the bandwidth the digital system can establish using each of these techniques, solely.

The main objective of this paper is to introduce various modulation techniques in order to have possibility of simultaneous transmission and multiplexing of signals.