本文選題：TLV320AIC23 + 語(yǔ)音算法�。� 參考：《成都理工大學(xué)》2013年碩士論文

【摘要】：音頻在現(xiàn)實(shí)生活中廣泛存在，對(duì)于音頻的處理很是重要。目前，在很多語(yǔ)音處理系統(tǒng)中都用到了語(yǔ)音分析模塊，采集現(xiàn)場(chǎng)的聲音并進(jìn)行頻譜分析。語(yǔ)音處理系統(tǒng)的實(shí)時(shí)性、功耗、體積、以及對(duì)語(yǔ)音信號(hào)的保真度都是很影響系統(tǒng)性能的關(guān)鍵因素。因此，音頻信號(hào)的分析器的設(shè)計(jì)是非常必要的。 隨著技術(shù)的進(jìn)步，數(shù)字化技術(shù)已經(jīng)深入人們的生活，數(shù)字化技術(shù)具有諸多模擬技術(shù)無(wú)可比擬的優(yōu)勢(shì)。傳統(tǒng)的音頻技術(shù)已經(jīng)不能滿足人們的要求，數(shù)字化的音頻處理方式成為了音頻技術(shù)的發(fā)展趨勢(shì)。數(shù)字技術(shù)的快速發(fā)展，，在很多場(chǎng)合使得數(shù)字音頻技術(shù)逐步取代模擬音頻技術(shù)，數(shù)字化的音頻處理是利用數(shù)字濾波算法對(duì)采集到的信號(hào)進(jìn)行變換處理來(lái)實(shí)現(xiàn)的。 采用數(shù)字化的技術(shù)，就必定會(huì)涉及到很多復(fù)雜的數(shù)字運(yùn)算，而數(shù)字運(yùn)算處理器（DSP）的出現(xiàn)和發(fā)展正是迎合了這一要求。DSP芯片的很多特殊結(jié)構(gòu)可以快速實(shí)現(xiàn)各種數(shù)字信號(hào)處理算法。 本文首先介紹了基于TMS320C5402DSP芯片的語(yǔ)音分析系統(tǒng)的工作原理，給出了整體設(shè)計(jì)方案和工作框圖，然后給出了系統(tǒng)的硬件設(shè)計(jì)方案；接著介紹了基于TMS320C5402DSP芯片的語(yǔ)音錄放系統(tǒng)的軟件設(shè)計(jì)。本文以小波變換及多分辨分析為理論基礎(chǔ)，對(duì)語(yǔ)音端點(diǎn)檢測(cè)中小波系數(shù)方差算法和子帶平均能量算法進(jìn)行了分析和研究，利用語(yǔ)音和噪聲的頻域差別，對(duì)這兩種算法進(jìn)行了優(yōu)化，并應(yīng)用于端點(diǎn)檢測(cè)系統(tǒng)中，有效地改善了小波系數(shù)方差算法耗時(shí)長(zhǎng)、實(shí)時(shí)性差的缺點(diǎn)，并克服了子帶平均能量算法只對(duì)高斯白噪聲檢測(cè)效果好的局限性，提高了語(yǔ)音端點(diǎn)檢測(cè)系統(tǒng)的實(shí)用性。 在整個(gè)設(shè)計(jì)過(guò)程中，我們采用了TLV320AIC23DSP芯片為核心音頻錄放接口器件，結(jié)合TMS320C5402DSP芯片，語(yǔ)音數(shù)據(jù)存儲(chǔ)FLASH存儲(chǔ)器等完成了數(shù)字音頻處理系統(tǒng)的設(shè)計(jì)。軟件部分則采用模塊化的設(shè)計(jì)方法，用C語(yǔ)言來(lái)實(shí)現(xiàn)。該數(shù)字音頻處理系統(tǒng)的設(shè)計(jì)具有以下特點(diǎn)： 1．音頻數(shù)據(jù)占用資源少。 2．音質(zhì)通信級(jí)。結(jié)合了音頻編碼與雙向通信所需的低延遲兩大優(yōu)勢(shì)，增強(qiáng)了話音編碼與優(yōu)質(zhì)音頻編碼的效果與質(zhì)量。 3．開(kāi)發(fā)難度低,僅通過(guò)TMS320C5402DSP芯片和幾個(gè)相關(guān)軟件（MATLAB和仿真軟件）即完成了相關(guān)設(shè)計(jì)。 4．語(yǔ)音芯片與DSP接口電路簡(jiǎn)單，利于生產(chǎn)和體積控制�？梢越档统杀荆⑶曳奖銛y帶。
[Abstract]:Audio is widely used in real life and is very important to audio processing. At present, voice analysis module is used in many speech processing systems. The real-time, power consumption, volume and fidelity of speech signal are the key factors that affect the performance of speech processing system. Therefore, the design of audio signal analyzer is very necessary. With the progress of technology, digital technology has been deep into people's lives, digital technology has many unparalleled advantages of analog technology. Traditional audio technology can not meet the needs of people, digital audio processing has become the development trend of audio technology. With the rapid development of digital technology, digital audio technology is gradually replacing analog audio technology in many situations. Digital audio processing is realized by using digital filtering algorithm to transform the collected signals. Using digital technology will inevitably involve a lot of complex digital operations, and the emergence and development of digital computing processor (DSP) meets this requirement. Many special structures of DSP chip can quickly realize various digital signal processing algorithms. This paper first introduces the working principle of speech analysis system based on TMS320C5402 DSP chip, gives the overall design scheme and working block diagram, and then gives the hardware design scheme of the system. Then the software design of speech recording and playback system based on TMS320C 5402 DSP is introduced. On the basis of wavelet transform and multi-resolution analysis, this paper analyzes and studies the variance algorithm of wavelet coefficients and subband average energy algorithm for speech endpoint detection, and makes use of the difference between speech and noise in frequency domain. The two algorithms are optimized and applied to the endpoint detection system, which effectively improves the shortcomings of the wavelet coefficient variance algorithm, such as long time consumption and poor real-time performance. It overcomes the limitation that the sub-band average energy algorithm is only good for Gao Si white noise detection and improves the practicability of speech endpoint detection system. In the whole design process, we used TLV320AIC23 DSP chip as the core audio recording and playback interface device, combined with TMS320C5402 DSP chip, voice data storage flash memory to complete the design of the digital audio processing system. In the software part, the modular design method is adopted, and C language is used to realize it. The design of the digital audio processing system has the following characteristics: 1. Audio data occupy less resources. 2. Audio quality communication level. Combining the advantages of audio coding and bidirectional communication with low delay, the effect and quality of voice coding and high quality audio coding are enhanced. 3. Only the TMS320C5402 DSP chip and several related software MATLAB and simulation software) are used to complete the related design. 4. The interface circuit between speech chip and DSP is simple, which is conducive to production and volume control. Can reduce the cost, and easy to carry.
【學(xué)位授予單位】：成都理工大學(xué)
【學(xué)位級(jí)別】：碩士
【學(xué)位授予年份】：2013
【分類號(hào)】：TN912.3;TP368.1

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