![]() The bit duration of each of the 8 bit is then-Įncoding process is the selection of and implementation of various available encoding format. This means that every 125 micro-second, 8bits representing the analog signal sample comes out of the quantizer. In PCM, 8 bits/sample are used and sampled at the rate of 8Khz sample/sec (i.e sampling rate=Fs=8Khz). At each sampling instant, portion of the analog PAM signal is quantized to the nearest level and corresponding binary bit values appears at the output. Each discrete level has a corresponding binary bit values. The sampled PAM signal is quantized into discrete level. As a result of this sampled PAM signal is produced at the output of the sample/hold circuit. The frequency of the clock is the sampling frequency that turns ON/Off the transistors. On one input port the signal from the filter is fed into the sample and hold circuit and another another clocked pulse input is fed from another input port. The sample and hold circuit consists of transistors that are turned on/off using periodic clocked pulse. The sampling is performed on the output of the filter using sample and hold circuit. ![]() This is done to avoid overlapping of samples and produce ISI(Inter-Symbol Interference) Effect. A-Law and m-law are two methods/ algorithm of non-uniform quantization.įiltering involves passing the analog signal through a filter called the anti-aliasing filter that allows selected portion of signal energy/ power to pass and stops other signal energy/ power to pass through the filter. In the non-uniform quantization process the samples are converted to digital bits with discrete level have non-uniform spacing between them. In the uniform quantization process the samples are converted to digital bits using discrete level that are uniformly spaced apart. The sampling and quantization process has also two methods-uniform and non uniform quantization. It is however easy to understand the PCM process by explaining them as separate steps. The Sampling and Quantization is performed by a single physical device called ADC(Analog to Digital Converter) and the process itself is called Analog to Digital conversion. The whole process of converting the analog signal to digital form involves 4 steps. The analog signal such as speech is usually the input signal. Bit rate is calculated by:Īs with sample rate, the higher the bit rate, the better quality of the recorded sound.The PCM(Pulse Code Modulation) refers to the process of converting analog signal. Bit rateīit rate is simply a measure of how much data is processed for each second of sound. Typical bit depths are 16 bit and 24 bit. Just as with images, the higher the bit depth, the more accurately a sound can be recorded, but the larger the file size. Bit depthīit depth refers to the number of bits used to record each sample. This is high enough for good sound quality while keeping file size down to sensible levels. An audio file is usually recorded at 44.1 kilohertz. As a result, sound files are often a compromise between quality and size of file. However, the higher the sample rate, the larger the resulting file. However, if the sample rate is doubled - twice as many samples in the same time period - the resulting representation would be closer: A sound wave plotted from 20 samples If the samples recorded above were plotted on a graph, the resulting representation of the sound wave would not be too accurate: A sound wave plotted from 10 samples The higher the sample rate, the closer the recorded signal is to the original. Sample rate is the number of samples recorded in any given period of time. This data is then stored in a file for later use. ![]() The sound recorded at each sample point is converted to its nearest numeric equivalent: Sample įor example, a sound wave like this can be sampled at each time sample point: An analogue-to-digital converter will capture a sound wave at regular time intervals. ![]()
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