1、数字音响的原理。核心提示：数字音响设备的工作原理主要是利用模拟信号变换数字信号的方法来进行工作的，转换的方法虽然有很多，但是为常用的还是脉冲编码调制的方式，这种方法是1937年A. H. 里福斯发明的，这种方法就是所谓的PCM。(Pulse Code Modulation)。 PCM方式是法国人 A. H. 里福斯于1937年发明的, 早已广泛应用于通信之中。 随着半导体技术的进步, 特别是发展到超大规模集成电路阶段后, PCM方式应用于音响领域, 并进入家庭成为现实。
1. The principle of digital audio. Core tip: the working principle of digital professional audio equipment mainly uses the method of converting analog signals into digital signals. Although there are many conversion methods, the commonly used method is pulse code modulation. This method was invented by A. h. rivers in 1937. This method is the so-called PCM. (Pulse Code Modulation)。 PCM mode was invented by French A. h. rivers in 1937 and has long been widely used in communication. With the progress of semiconductor technology, especially after the development of VLSI, PCM mode has been applied to the field of sound and entered the family.
2、数字音响设备的基本组成。PCM方式是由取样, 量化和编码三个基本环节完成的。
2. Basic composition of digital audio equipment. PCM mode is completed by three basic links: sampling, quantization and coding.
3、数字音响设备的工作原理
3. Working principle of digital audio equipment
(1)取样。对振幅随时间连续变化的信号波形按一定的时间间隔取出样值, 形成在时间上不连续的脉冲序列, 称之为取样。这个时间间隔称为取样周期, 记为Ts, 相应的取样频率fs=1/Ts；
(1) Sampling. For the signal waveform whose amplitude changes continuously with time, the sample value is taken according to a certain time interval to form a time discontinuous pulse sequence, which is called sampling. This time interval is called the sampling period, which is recorded as ts, and the corresponding sampling
frequency FS = 1 / TS;
(2)量化。将模拟信号的幅度动态范围划分为相等间隔的若干层次, 把取样输出的信号电平按照四舍五入的原则归入靠近的量值, 称之为量化；
(2) Quantification. The amplitude dynamic range of analog signal is divided into several levels with equal intervals, and the signal level of sampling output is classified into the close value according to the principle of rounding, which is called quantization;
(3)编码。把取样, 量化所得的量值变换为二进制数码的过程称为编码。 在数字音响中, 通常采用16位(bit)数码表示一个量值, 即量化位数n=16；
(3) Code. The process of transforming the quantity obtained by sampling and quantization into binary code is called coding. In digital audio, 16 bit numbers are usually used to represent a quantity value, that is, the quantization bits n = 16;
(4)纠错编码。由于激光唱片和盒式磁带在制作和使用过程中会发生超过容许范围的损伤, 使所读出的数字信号与原来所记录的信号有所差别, 因此, 必须采取纠正错码的措施；
(4) Error correction coding. Because the damage beyond the allowable range will occur in the production and use of laser records and cassette tapes, so that the digital signals read out are different from the original recorded signals, so measures must be taken to correct the wrong code;
(5)帧结构。数字信号是以字符为单位的, 若偏移 1 位, 就会使该字符代表的信号电平发生变化。 为此, 必须把记录信号分割成很小的字组, 并设法判断出各字组之间的分界线, 这样的字组称为帧；
(5) Frame structure. Digital signal is in character unit. If offset by 1 bit, the signal level represented by this character will change. Therefore, the recorded signal must be divided into small word groups, and try to judge the dividing line between each word group. Such word group is called frame;
(6)调制。模拟音频信号经取样, 量化, 编码和CIRC纠错编码后形成的数字信号, 还不宜直接记录在唱片或磁带上。 因为在数据流中可能会出现 16 位全部为 0 或 1 的情况, 从唱片或磁带上读取时会使信号极不稳定, 也会造成伺服系统的不稳定。
(6) Modulation. The digital signal formed by sampling, quantization, coding and CIRC error correction coding of analog audio signal should not be directly recorded on record or tape. Because all 16 bits may be 0 or 1 in the data stream, the signal will be extremely unstable when reading from the record or tape, and it will also cause the instability of the servo system.
以上内容就是今天小编为您介绍的
山东舞台音响的相关知识，更多的内容请继续关注我们。
The above content is the Shandong stage audio system introduced to you by Xiaobian today For more information, please continue to pay attention to us.
