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Chapter 10: Source Coding
Digitization steps for analog signals? Its details?
Sampling, quantization, encoding.
The digitization of analog signals is to convert analog signals into digital signals, so that the purpose of transmitting signals in digital transmission systems can be achieved.
Sampling is to sample the analog signal according to the sampling theorem, and convert the analog signal into a sampling signal. There are two types of low-pass analog signal and loan-pass analog signal. The former requires sampling frequency Fs≥2f; the latter requires Fs≥2B(1+k/n), k is the fractional part of FH (the highest band-pass frequency)/B, and n is the integer part.
Sampling modulation with periodic pulses. A periodic pulse has four variable parameters: period, amplitude, width, and phase. Periods generally follow the sampling theorem. Therefore, there are three modulation methods: PAM, PWM, and PPM. Generally, PAM is used to control the amplitude for sampling modulation.
Quantization is to convert the sampling signal into a quantized signal, which can be divided into two types: uniform quantization and non-uniform quantization. The number of quantization levels and the quantization interval of uniform quantization are fixed, while the quantization interval of non-uniform quantization varies with the sampling value of the signal. The advantage of non-uniform quantization is that it occupies fewer bits than uniform quantization. Non-uniform quantization has A compression ratio and μ compression ratio. The implementation methods include 13 fold lines and 15 fold line compression methods. Its code composition is polar code+paragraph code+intra-segment code. Features: The paragraphs are uneven, but the paragraphs are uniform.
Encoding is a quantized signal represented by one or more bits of binary. There are two commonly used encodings: natural binary and folded binary. So far, an analog signal is represented by a digital signal, and then the signal can be transmitted through a digital baseband system or a digital bandpass system, and finally the signal is recovered and the analog signal is decoded.
Such a process is called pulse code modulation (PCM). However, the coding efficiency of PCM is not high and occupies a large bandwidth, so there is differential pulse code modulation (DPCM). It is to reduce the transmission bit of the digital signal and reduce the bandwidth occupation. The idea of difference is to calculate a predicted value based on the previous sampled value, and then use the difference between the current sampled value and the predicted value as coding. Since there is a certain correlation between the continuous values of the speech signal, the difference made changes The amplitude will not be particularly large, so the overall transmission bit rate will be reduced. Incremental modulation is the simplest DPCM, because its quantization level is 2, and there are only two states of 0 and 1.
What is the purpose of source compression coding?
The purpose of signal source coding is to reduce signal redundancy and improve signal effectiveness. Generally speaking, the coding part of the information source is compressed symbols, and the number of symbols is reduced as much as possible. For different signals, coding has different requirements. Like voice signals and most image signals, lossy compression is generally used; while data compression and medical image compression must be lossless compression.