PCM Audio Coding Part 2
Coding
The quantized sampled signal is converted into a series of decimal digital code streams arranged according to the sampling sequence, ie, a decimal digital signal. A simple and efficient data system is a binary code system. Therefore, the decimal digital code must be converted to a binary code. Based on the total number of decimal digital codes, the number of binary code bits required can be determined, that is, the word length (sampling bits) The process of transforming the quantized sample signal into a binary code stream of a given word length is called encoding.
Example
Next, the above 1.65 V corresponds to a quantization level of 128. The corresponding binary system is 10000000. That is, the encoded result of the sample point is 10000000. Of course, this is an encoding method without considering values positive and negative, and there are many types of coding methods that require specific analysis of specific problems. (PCM audio format encoding is A-law 13 polyline encoding)
PCM audio encoding
PCM signals are not subject to any encoding or compression (lossless compression). Compared to analog signals, it is less susceptible to transmission system clutter and distortion. The dynamic range is wide and the sound quality is quite good. The coding adopts the A-law 13 polyline coding.
A-grade 13-fold line
The A-law is a form of logarithmic compansion in PCM non-uniform quantization. Digital pulse code modulation (PCM) is the basic method for digitizing analog signals today. PCM includes three steps: sampling, quantization, and encoding. Among them, the quantization is the discrete value of the sampling values. Uniform quantization and non-uniform quantization can effectively improve the quantization signal-to-noise ratio of the signal. Quantization of speech signals often takes two logarithmic forms of non-uniform quantization and ITU-recommended compression characteristics: A-law and Mu-law. A-law coding is primarily used in 30/32-channel group systems, and A- The PCM law is used in Europe and China.
See the article for more details
channel
Channels can be divided into PCM mono and stereo (two channels)
.Each sample value is contained in an integer i whose length is the minimum number of bytes needed to accommodate the specified sample length.
The low significant byte is stored first, and the bit representing the sample amplitude is placed in the high significant bit of i, and the remaining positions are 0, so the data format of the PCM waveform samples 8 and 16 bit is as follows.
