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CP5010_Essay_MINIBLOCK_2(2) - Essay Example

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PCM Theory and Audio Reduction Codecs and Techniques Name Institution Abstract Pulse-code modulation or PCM is a digital exemplification of an analog signal whereby the signal magnitude is sampled regularly at even intervals, then quantized to series of symbols in digital (usually binary) code…
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Download file to see previous pages... Instead, compressed PCM variants are normally employed. However, numerous Blu-ray format movies use the uncompressed PCM for audio. Frequently, PCM encoding enables digital transmission from a point to another (in a certain system, or geographically) in a serial form. This paper evaluates PCM theory and audio reduction codecs and techniques. Introduction In the past, the communication system mostly used the analog signal in transmitting signals. However, due to computer and digital network communications advancements, a lot of information or data is transmitted using the pulse wave modulation technique. Pulse wave modulation may be used to transmit analog audio signal or information with a particular rate to sample analog signal - this rate is what is called the transmission rate. On the receiver, the delivered signal is demodulated by the Pulse-code modulation (PCM) demodulator to recuperate the original continuous analog signal wave. Generally, PCM can be classified as a pulse amplitude modulation (PAM), pulse position modulation (PPM), pulse code modulation (PCM), and pulse width modulation (PWM). PAM, PPM, and PWM modulations are affiliated to analog modulation while the PCM modulation is affiliated to digital modulation. It is important to take note that PCM modulation is real digital signal which can be processed and digitally stored by computer. However, PPM, PWM, and PAM modulations are similar to PM, FM, and AM modulations, respectively (Aksoy & DeNardis, 2007, p. 112). For all pulse wave modulation, before the modulation, the original continuous form signal have to be sampled and the sampling rate for the sampling signal must not be low, otherwise the recovered signal will bring about distortion. The sampling rate is subject to the sampling theorem, whereby the sampling theorem states that: for a pulse wave modulation system, in condition that the sampling rate excesses the double or more maximum frequency times of the signal, then distortion level of data recovery on the receiver will be its minimum. For instance, the frequency range of audio signal is about 40 Hz ~ 4 kHz, the pulse wave modulation sampling signal frequency must be no less than 8 kHz, hence, the sampling error is reduced to minimum (Maes & Vercammen, 2012, p. 67). During transmission, it is hard for the PCM signal to avoid the noise distortion. Therefore, before PCM signal sends to the PCM demodulator, a comparator is used to recover the PCM signal to its original level. The signal is a pulse wave signal series, so, before demodulating, the pulse wave signal series will be converted to a parallel digital signal by aid of a serial to a parallel converter After that, the signal passes through n-bits decoder (which should be D/A converter) for recovery of the digital signal to its original quantization value. However, in this quantization value includes not only the original audio signal alone, but also many high frequency harmonics, hence; a low-pass filter is used to remove the undesirable signal at the final fragment. MPEG varieties The MPEG standards comprise of different Parts. Every part covers a certain element of the entire specification. The standards also specify Levels and Profiles. Profiles are intended to delineate a set of available tools, and Levels delineate the range of suitable values for the ...Download file to see next pagesRead More
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