微控制器的音頻編碼與壓縮

　　將語(yǔ)音和聲音生成添加到產(chǎn)品中可以大大提高產(chǎn)品的可用性和可銷(xiāo)售性，并且不需要添加單獨(dú)的數(shù)字信號(hào)處理器（DSP）或?qū)ｉT(mén)的音頻處理器來(lái)實(shí)現(xiàn)。微控制器（MCU）廠商已經(jīng)實(shí)現(xiàn)了DSP擴(kuò)展，使那些曾經(jīng)被認(rèn)為太復(fù)雜的算法在一個(gè)專(zhuān)用DSP平臺(tái)上實(shí)現(xiàn)了實(shí)時(shí)解碼。

　　此外，音頻處理提供了大量的權(quán)衡，可以利用設(shè)計(jì)師允許復(fù)雜的音頻是在相對(duì)低速的8位微控制器。例如，它可以實(shí)現(xiàn)標(biāo)準(zhǔn)的自適應(yīng)差分脈沖編碼調(diào)制（ADPCM）簡(jiǎn)單的版本，在一個(gè)相對(duì)簡(jiǎn)單的單片機(jī)的算法，如Microchip PIC16不要求任何DSP擴(kuò)展。

　　為減小存儲(chǔ)容量而設(shè)計(jì)的音頻編碼技術(shù)分為兩類(lèi)。第一種方法稱(chēng)為波形編碼，它使用波形本身的已知屬性。波形編碼的優(yōu)點(diǎn)是它試圖對(duì)信號(hào)進(jìn)行編碼，而不需要知道信號(hào)是如何產(chǎn)生的。這使得編解碼器可以應(yīng)用于不同形式的音頻，即使該算法可能是為例如語(yǔ)音而設(shè)計(jì)的。

　　Speech-oriented waveform codecs often take advantage of the observation that there is a higher probability of a speech signal taking a small value rather than a large value. Therefore， a speech processor can reduce the bit rate by quantizing the smaller samples with finer step sizes and the large samples with coarse step sizes. The bit rate can further be reduced by using one inherent characteristic of speech： a high correlation between consecutive speech samples.

　　Rather than encode the speech signal itself， the difference between consecutive samples can be encoded. This method is a relatively simple one that is repeated on each sample with little overhead from one sample to the next. ADPCM is an example of a waveform algorithm that uses this technique.