基于DSP的语音编码中的模糊增量调制算法的实现
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摘要
随着多媒体通信不断地融入到了我们的生活中,语音编码扮演着越来越重要的角色。目前应用比较广泛的有波形编码方式以及参数编码和混合编码方式,虽然参数编码和混合编码在编码速率上有一定的优势,但因为其解码恢复出来的语音自然度较低,且算法复杂度较高,对实时处理有一定的局限,而波形编码以其硬件实现电路简单,具有较强的抗干扰能力及极强的保密性仍广泛应用于军事通信等领域。目前广泛采用的算法有差分编码调制(DPCM),自适应差分编码调制(adaptive DPCM),增量调制(△M)。其主要原理都是对预测器的预测值与实际值之差编码,其预测器中Δ的变化都是依赖于精确的数学模型,这使得对于一些幅度变化特别大的信号来说仍然会带来严重的颗粒噪声或斜率过载失真。
     为解决上述问题,在本设计中将模糊控制理论引入到增量调制中,提出了一种模糊增量调制算法,它具有无需确定其准确数学模型的特点,即将预测值与实际值之差e(n)及e(n)与e(n-1)之差Δe(n)作为输入,经过模糊控制器一系列推理输出精确的控制量δ。最后通过合理选取参数初始值,在matlab上对此算法进行仿真并分析其性能,为了进一步考查此算法的实时处理能力,将此算法移植到具有强大的数字信号处理能力的DSP芯片TMS320C5402的开发板上,结果表明在DSP上有较好的编解码效果:量阶δ的大小能够随着语音信号幅度的变化快慢自动地改变,使预测器输出能更好地跟踪快速变化的信号波形,既具备了ADM的优点,又比它有更好的SNR性能和更大的动态范围,基本实现了“透明”编码,达到了设计要求。
With the integration of multimedia communications to keep our life, speech coding plays an increasingly important role. At present, there is a broader application of waveform coding and the parameter coding and hybrid coding, although parameters of coding and hybrid coding have on the advantages at coding rates, but for its decoding restored voice's nature is low and high complexity algorithm for real-time processing so they have limitations, for waveform coding circuit has easy hardware implementation, strong interference capability and Security, so it is still very widely used in military communications and other fields, now there is a widely used algorithms such as DPCM, ADPCM,ΔM. the main principle of it is to code with the difference between forcasting value and the actual value .the changes ofΔin the predictor are dependent on the precise mathematical model . which still causes severe slope overload noise or distortion for some particularly high rate of change of the signal.
     To address these issues, come up with an algorithm—fuzzy delta modulation algorithm using fuzzy control theory, which needn't to determine the accurate mathematical model of the process, that is put the difference between Predictive value and the actual value as input , then by application of fuzzy control theory to predict the difference (e(n))between the value of prediction and the actual value and the difference (Δe (n)) between e (n) and e (n-1) in fuzzy way , get a a series of fuzzy control through fuzzy inference based on fuzzy control rules .then get precise value of control output ofδthrough fuzzy theory. Select the parameters of the initial adoption of a reasonable value, simulate and analysis of its performance on matlab, in order to further test this algorithm in real-time processing capability, at last migrate this algorithm to a powerful digital signal processing capability of DSP chip TMS320C5402 development board as the core, the results show that this algorithm in the DSP codec better effect: the volume ofδsize changes automatically with the voice signal amplitude changes which make the forecast better track the rapid changes in the signal waveform, it has not only the advantage of ADM, but also a better SNR performance and more dynamic the scope, fulfill the basic realization of the 'transparent' coding, then achieve the design requirements.
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