1、數字音響的原理。核心提示：數字音響設備的工作原理主要是利用模擬信號變換數字信號的方法來進行工作的，轉換的方法雖然有很多，但是為常用的還是脈沖編碼調制的方式，這種方法是1937年A. H. 里福斯發明的，這種方法就是所謂的PCM。(Pulse Code Modulation)。 PCM方式是法國人 A. H. 里福斯于1937年發明的, 早已廣泛應用于通信之中。 隨著半導體技術的進步, 特別是發展到超大規模集成電路階段后, PCM方式應用于音響領域, 并進入家庭成為現實。
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、數字音響設備的基本組成。PCM方式是由取樣, 量化和編碼三個基本環節完成的。
2. Basic composition of digital audio equipment. PCM mode is completed by three basic links: sampling, quantization and coding.
3、數字音響設備的工作原理
3. Working principle of digital audio equipment
(1)取樣。對振幅隨時間連續變化的信號波形按一定的時間間隔取出樣值, 形成在時間上不連續的脈沖序列, 稱之為取樣。這個時間間隔稱為取樣周期, 記為Ts, 相應的取樣頻率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。將模擬信號的幅度動態范圍劃分為相等間隔的若干層次, 把取樣輸出的信號電平按照四舍五入的原則歸入靠近的量值, 稱之為量化；
(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)編碼。把取樣, 量化所得的量值變換為二進制數碼的過程稱為編碼。 在數字音響中, 通常采用16位(bit)數碼表示一個量值, 即量化位數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)糾錯編碼。由于激光唱片和盒式磁帶在制作和使用過程中會發生超過容許范圍的損傷, 使所讀出的數字信號與原來所記錄的信號有所差別, 因此, 必須采取糾正錯碼的措施；
(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)幀結構。數字信號是以字符為單位的, 若偏移 1 位, 就會使該字符代表的信號電平發生變化。 為此, 必須把記錄信號分割成很小的字組, 并設法判斷出各字組之間的分界線, 這樣的字組稱為幀；
(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)調制。模擬音頻信號經取樣, 量化, 編碼和CIRC糾錯編碼后形成的數字信號, 還不宜直接記錄在唱片或磁帶上。 因為在數據流中可能會出現 16 位全部為 0 或 1 的情況, 從唱片或磁帶上讀取時會使信號極不穩定, 也會造成伺服系統的不穩定。
(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.
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