progress bar code in vb.net Figure 3-74 PAM samples. in Software

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Figure 3-74 PAM samples.
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Figure 3-75 Quantizing the samples.
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noise and is inevitable when linear measurement systems, such as the one suggested by Figure 3-75, are employed in CODECs. Needless to say, design engineers recognized this problem rather quickly, and equally quickly came up with an adequate solution. It is a fairly well-known fact among psycholinguists and speech therapists that the human ear is far more sensitive to discrete changes in amplitude at low volume levels than it is at high volume levels, a fact not missed by the network designers tasked with optimizing the performance of digital carrier systems intended for voice transport. Instead of using a linear scale for digitally encoding the PAM samples, they designed and employed a nonlinear scale that is weighted with much more granularity at low volume levels that is, close to the zero line than at the higher amplitude levels. In other words, the values are extremely close together near the x-axis, and get farther and farther apart as they travel up and down the y-axis. This nonlinear approach keeps the quantizing noise to a minimum at the low amplitude levels where hearing sensitivity is the highest and enables it to creep up at the higher amplitudes, where the human ear is less sensitive to its presence. It turns out that this is not a problem, because the inherent shortcomings of the mechanical equipment (microphones, speakers, the circuit itself) introduce slight distortions at high amplitude levels that hide the effect of the nonlinear quantizing scale. This technique of compressing the values of the PAM samples to make them fit the nonlinear quantizing scale results in a bandwidth savings of more than 30 percent. In fact, the actual process is called companding, because the sample is first compressed for transmission, and then expanded for reception at the far end, hence the term.
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The actual graph scale is divided into 255 distinct values above and below the zero line. In North America and Japan, the encoding scheme is known as -Law (Mu-Law). The rest of the world relies on a slightly different standard known as A-Law. Eight segments can be found above the line and eight below (one of which is the shared zero point). Each segment, in turn, is subdivided into 16 steps. A bit of binary mathematics now enables us to convert the quantized amplitude samples into an 8-bit value for transmission. For the sake of demonstration, let s consider a negative sample that falls into the thirteenth step in segment five. The conversion would take on the following representation: 1 101 1101 where the initial 0 indicates a negative sample, 101 indicates the fifth segment, and 1101 indicates the thirteenth step in the segment. We now have an 8-bit representation of an analog amplitude sample that can be transmitted across a digital network and then be reconstructed with its many counterparts as an accurate representation of the original analog waveform at the receiving end. This entire process is known as PCM, and the result of its efforts is often referred to as toll-quality voice.
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Alternative Digitization Techniques
Although PCM is perhaps the best-known, high-quality voice digitization process, it is by no means the only one. Advances in coding schemes and improvements in the overall quality of the telephone network have made it possible for encoding schemes to be developed that use far less bandwidth that traditional PCM. In this next section, we will consider some of these techniques.
Adaptive Differential Pulse Code Modulation (ADPCM)
Adaptive Differential Pulse Code Modulation (ADPCM) is a technique that enables toll-quality voice signals to be encoded at a half rate (32 Kbps) for transmission. ADPCM relies on the predictability that is
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