!244269 玖、發明說明： 【發明所屬之技術領域】 本發明係關於一種數位音頻優化器，特別是指一種透過 運算方式，使其可輸出平坦之頻率響應曲線之數位音頻優化 器。 【先前技術】 習用為解決因喇A本體頻率響應曲線的不良，再加上音 箱所造成之第二次頻率響應曲線不良的缺失，通常係於電路 中加入補償電路，以進行喇队及音箱頻率響應曲線的調整， 而習用之補償電路之構成如下所述·· 一、習用補償電路，係使用多數被動元件，例如：電感、 電容及電阻所組成之低通濾波器、高通濾波器、帶通濾波器 等夕種型式濾波器，以串聯或並聯等排列組合方式，作衰減 或增加等方式來解決頻率響應不良。然而，上述方式雖可改 善頻率響應不良之缺失，4旦因其係將電感 '電容、電組排列 組合，一旦頻率曲線中有更多頻率點需要作修正時，所造成 之困難度相對高，且所需之元件數亦相對增加。因此，當體 積增加時，成本即相對提#，且此方式通常只做部份之修 正’並無法做到全面之修正。 一砷汉吾上述I用補償電路因使用之元件體積較大 造成整體電路體積相對增加之缺失，故以放大器來替代滤 器，。其财在前級放大器中，以放大器之特性設計為主動; 大器之南通放大器、低通放Ali、帶通放大器等運用放大 放大倍率之射生，作適度"+、"方且必須係以多組作串, 式或並聯式等排肋合方式取得所f的頻率曲線 q 用多級放大器線路所需之元件比濾、波器之運用元 小’但其所使用之7C件個數卻更多，若在所需調整之頻率舅 1244269 愈夕時，將導致所需之級數就愈多，如此，不但造成體積及 成本之增加外，另外還造成失真度相對增加，使其在設計上 之困難度隨之提昇。 再者，以濾波器或放大器做為補償電路，其皆具有一共 同之缺點，即喇，八本體改變後，頻率響應曲線一定會改變， 而音箱設計及製造有所變更時，其頻率響應曲線也會跟著改 變二因此，當喇叭及音箱有所變更時，其必須將所有線路重 新。又计 旦剩p八或音相無法作任何變更時，將造成頻率燮 應曲線無法作調整，此為設計上之重大瓶頸，成頻羊0 由此可見，上述習用物品仍有諸多缺失，實非一良善之 设计者’而巫待加以改良。 本案發明人鑑於上述習用補償電路所衍生的各項缺點， 乃亟思加以改良創新，並經多年苦心孤詣潛心研究後，終於 成功研發完成本件數位音頻優化器。 【發明内容】 本發明之目的即在於提供一種可隨時將所取得之喇。八及 音箱的頻率響應曲線在電腦上先做適度模擬調整後，再將此 曲線資料作儲存，以得到最佳頻㈣應曲線之數位音頻優化 器。 本發月之-人目的係在於提供一種在製造過程中，只須 將儲存有喇八及音箱頻率響應曲線之唯讀記憶體插到電路 板上gp可几成隨時變更設計目的之數位音頻優化器。 本卷明之另一目的係在於提供一種透過CPU將DSP運算 、、。果轉換成串列信號，再傳送至數位/類比信號轉換器轉換 成類比㈣，以減少·在作複雜運算時所需的負擔，進而 達到維濩聲音品質目的之數位音頻優化器。 本發明之又一目的係在於提供一種先行設定數位/類比 1244269 信號轉換器之輪出最大位準，以預防功率放大器因過大信號 矜夺這成切割失真之現象，使其無須像傳統方式使用多 數兀件及複合線路組成一壓縮線路來改善切割失真之數位音頻 優化器。 丄y達成上述發明目的之數位音頻優化器，包括有類比/數 位k唬轉換器、CPU、數位信號處理器（DSP)、唯讀記憶體、 數=/類比信號轉換器、濾波器/緩衝器、功率放大器、喇口八 及音箱；其中，該唯讀記憶體係可供喇π八及音箱之頻率響應 曲線儲存；該類比/數位信號轉換器係接收音頻信號，並將該 音頻信號轉換成數位信號後，再傳送至CPU，該CPU會將所 接收之佗號轉換成並列信號後，再將信號送往DSP與唯讀記 憶體内儲存之頻率曲線作運算，再將運算結果送回^PU轉成 串列信號，再傳送至數位/類比信號轉換器轉換成類比信號， 再經由濾波器及緩衝器（FILTER&BUFFER)進行濾波及功率放 大器作功率放大後，再輸出喇A及音箱，使其可輸出平坦之 頻率響應曲線。 【實施方式】 請參閱圖一所示，係本發明所提供之數位音頻優化器， 主要包括有： ° 一類比/數位信號轉換器i，其係接收來音響之音頻類比 信號10,並將該音頻類比信號10轉換成數位串列信號n後， 傳送至CPU2 ; 一 CPU2，該CPU2係為一控制中心，其係接收來自類比/ 數位信號轉換器1之數位串列信號U，並將該數位$列信號 11排列成並列信號21後，再將該信號21傳送至數位信號處 理器3 ( DSP)作運算，運算完成，再將運算完成之參數信號 送回CPU2轉成數位串列信號22後，傳送至數位/類比信號轉 1244269 換器4 ; 一唯讀記憶體5 ( EEPROM)，該唯讀記憶體$係與數位信 號處理器4介接，其主要係儲存喇队8及音箱9原有之頻率 響應曲線爹數； 一數位仏唬處理器3 ( DSP)，該數位信號處理器3係接收 來自CPU2及唯讀記憶體5之信號，並依所接收之信號進行運 算，其係以喇π八之頻率響應曲線作為運算參數，例如·· a信 號參數值+B信號參數值信號參數值，A信號係代表喇= 8及音箱9之頻率響應曲線，B信號則等於A信號之負數，c 信號為平坦之頻率響應曲線，因此，為使喇W 8及音箱9可 輸出平坦曲線之C信號，該數位信號處理器3於運算時，須 先擷取儲存在唯讀記憶體5中之A信號，以得到該a信號之 負數B信號參數，並將該負數B信號參數傳送回cpu2轉換回 數位串列信號22後，再傳送至數位/類比信號轉換器4，如此 傳送方式，將可降低數位信號處理器3在作複雜運算時之負 擔，以避免降低傳送速度，導致最後之聲音品質相對降低； 數位/類比彳s號轉換器4，其係接收來自cpU2之負數b L號將所接收之佗號轉換成類比信號41後，再傳送至濾波 器緩衝器6中進行濾波緩衝後，再傳送至功率放大器7作功 率放大後帛後’將放大之負數B信號傳送至_ ^ 8及音箱 9中，使得喇< 8及音箱9之頻率響應曲線之A信號參數加負 數之B信號參數可得到c信號，該c信號會等於” 〇”，使得喇 队8及音箱9可輸出平坦之頻率響應曲線；另外，cpu2並會 數位仏號處理器3之運算結果，設定數位/類比信號轉換器 4士之輸出最大位準，以預防在功率放大器7因過大信號輸入 時’所造成之切割失真。 另外，該類比/數位信號轉換器4經CPU2所取得之立體 1244269 音頻信號資料，可經由數位信號處理器3將低通濾波器（L〇w PASS FILTER)及高通濾波器（HIGH PASS FILTER)之左、右兩 聲道相加再除以二，同時再加上低通濾波器，再將此信號與 另一喇叭參數作運算後，再經由另一數位/類比信號轉換°器 4 ’如此又可得到另一聲道（重低音）良好曲線之輪出。 再者，將喇队8及音箱9原有之頻率響應曲線參數儲存 在唯凟§己憶體5或相關類似元件中，一旦剩w 8及音箱9之 頻率響應曲線改變時，該改變之頻率響應曲線參數將可隨時 儲存在唯讀記憶體5或相關類似之元件中，以改善習用補償 電路之缺失。 本發明所提供之數位音頻優化器，與其他習用技術相互 比較時’更具有下列之優點： 1.本發明可隨時將所取得之喇p八及音箱的頻率響應曲線 在電腦上先做適度模擬調整後，再將此曲線資料作儲存，以 得到最佳之頻率響應曲線。 2·本發明在製造過程中，只須將儲存有喇 < 及音箱頻率響 應曲線之唯讀記憶體插到電路板上，即可完成隨時變更設計 之目的。 ° ° 3·本發明係透過cpu將DSp運算結果轉換成串列信號，再 傳送至數位/類比信號轉換器轉換成類比信號，以減少"^^^ 在作複雜運算時所需的負擔，進而達到維護聲音品質之目 的。 4.本發明係先設定數位/類比信號轉換器之輸出最大位 準，以預防功率放大器因過大信號輸入時，造成切割失真之 現象，使其無須像傳統方式使用多數元件及複合線路組成一壓 縮線路來改善切割失真。 上列詳細說明係針對本發明之一可行實施例之具體說 1244269 列並非用以限制本發明之專利範圍，凡未脫離 X Α技#精神所為之等效實施或變更，均應包含於本案之 專利範圍中。 /、 练上所述，本案不但在技術構思上確屬創新，並能較習 用2 增進上述多項功效，應已充分符合新穎性及進步性之 疋發月專利要件，爱依法提出申請，懇言青貴局核准本件 發明專利申請案，以勵發明，至感德便。 【圖式簡單說明】 π參閱以下有關本發明一較佳實施例之詳細說明及其附 圖，將可進一步瞭解本發明之技術内容及其目的功效；有關 該實施例之附圖為： 圖一為本發明數位音頻優化器之方塊圖。 【主要部分代表符號】 1 類比/數位信號轉換器 10 音頻類比信號 11 數位串列信號 2 CPU 21 並列信號 22 數位串列信號 3 數位信號處理器 4 數位/類比信號轉換器 41 類比信號 5 唯讀記憶體 6 濾波器/緩衝器 7 功率放大器 8 制口八244269 发明 Description of the invention: [Technical field to which the invention belongs] The present invention relates to a digital audio optimizer, and more particularly to a digital audio optimizer that can output a flat frequency response curve through an arithmetic method. [Previous technology] Conventionally, in order to solve the poor frequency response curve of the La A body, and the lack of the second poor frequency response curve caused by the speaker, a compensation circuit is usually added to the circuit to perform the frequency of the speaker and the speaker. The response curve is adjusted, and the structure of the conventional compensation circuit is as follows: 1. The conventional compensation circuit uses most passive components, such as low-pass filters, high-pass filters, and band-passes composed of inductors, capacitors, and resistors. Various types of filters, such as filters, are arranged and combined in series or parallel, and attenuated or increased to solve the poor frequency response. However, although the above method can improve the lack of poor frequency response, because it is a combination of inductors, capacitors, and electric groups, once more frequency points in the frequency curve need to be corrected, the difficulty is relatively high. And the number of components required is relatively increased. Therefore, when the volume is increased, the cost is relatively increased, and this method usually only partially corrects it, and it cannot complete the correction. Since the above-mentioned compensation circuit for I used is relatively large due to the large volume of components used, the overall circuit volume is relatively increased, so an amplifier is used instead of the filter. Its wealth is in the pre-amplifier, taking the characteristics of the amplifier as the active design; large-scale Nantong amplifier, low-pass amplifier Ali, band-pass amplifier and other applications using amplification magnification, for moderate " +, " Multi-groups are used to obtain the frequency curve of f in a series, parallel or parallel rib arrangement. The components required for the multi-stage amplifier circuit are smaller than the operating elements of filters and wave filters. However, the 7C pieces used are But the number is more. If the frequency to be adjusted is 舅 1244269, it will lead to more stages. In this way, not only will it increase the volume and cost, but it will also cause a relative increase in distortion, making it Difficulties in design have increased. In addition, the use of filters or amplifiers as compensation circuits has a common disadvantage, that is, after the eight body is changed, the frequency response curve must change, and when the speaker design and manufacturing are changed, the frequency response curve It will also follow change two. Therefore, when the speakers and speakers are changed, they must reconnect all lines. In addition, if the remaining p-eight or the sound cannot be changed in any way, the frequency response curve cannot be adjusted. This is a major bottleneck in the design. It can be seen from this that the conventional items still have many defects. Not a good designer 'and the witch needs to be improved. In view of the various shortcomings derived from the conventional compensation circuit, the inventor of this case has been eager to improve and innovate. After years of painstaking research, he finally successfully developed this digital audio optimizer. [Summary of the Invention] The object of the present invention is to provide a device that can be obtained at any time. The frequency response curve of the eighth speaker is adjusted on the computer after appropriate analog adjustment, and then the curve data is stored to obtain a digital audio optimizer with the best frequency response curve. The purpose of this month is to provide a digital audio optimization that can be changed at any time during the manufacturing process by simply inserting the read-only memory that stores the frequency response curve of the speaker and the speaker into the circuit board. Gp can be changed at any time. Device. Another purpose of this volume is to provide a kind of DSP operation through CPU. A digital audio optimizer that converts serial signals to digital / analog signal converters and converts them to analog signals to reduce the burden required for complex calculations and to maintain the sound quality. Another object of the present invention is to provide a preset digital / analog 1244269 signal converter maximum level to prevent the power amplifier from seizing the cutting distortion due to excessive signal, so that it does not need to use most of the traditional methods. Digital components and composite circuits form a compressed audio digital optimizer to improve cutting distortion.音频 y A digital audio optimizer that achieves the above-mentioned object of the invention includes an analog / digital converter, a CPU, a digital signal processor (DSP), a read-only memory, a digital / analog converter, and a filter / buffer. , Power amplifier, Lakou eight and speaker; Among them, the read-only memory system can store the frequency response curve of La π eight and speaker; the analog / digital signal converter receives the audio signal and converts the audio signal into digital After the signal is sent to the CPU, the CPU will convert the received signal into a parallel signal, and then send the signal to the DSP and the frequency curve stored in the read-only memory for calculation, and then return the calculation result to ^ PU It is converted into a serial signal, and then transmitted to a digital / analog signal converter to be converted into an analog signal, and then filtered and filtered by a filter and a buffer (FILTER & BUFFER), and the power amplifier is used for power amplification. It can output a flat frequency response curve. [Embodiment] Please refer to FIG. 1, which is a digital audio optimizer provided by the present invention, which mainly includes: ° An analog / digital signal converter i, which receives an audio analog signal 10 from a sound, and After the audio analog signal 10 is converted into a digital serial signal n, it is transmitted to CPU2; a CPU2, which is a control center, which receives the digital serial signal U from the analog / digital signal converter 1, and sends the digital After the column signal 11 is arranged into a parallel signal 21, the signal 21 is transmitted to the digital signal processor 3 (DSP) for calculation. After the operation is completed, the parameter signal of the operation is sent back to the CPU 2 and converted into a digital serial signal 22. , To the digital / analog signal to 1244269 converter 4; a read-only memory 5 (EEPROM), the read-only memory $ is connected to the digital signal processor 4, which mainly stores the original team 8 and the speaker 9 Some frequency response curve dad; a digital bluff processor 3 (DSP), the digital signal processor 3 receives signals from the CPU2 and the read-only memory 5, and performs operations according to the received signals, which is based on Frequency response The curve is used as a calculation parameter, for example, a signal parameter value + B signal parameter value signal parameter value, A signal represents the frequency response curve of La = 8 and speaker 9, B signal is equal to the negative number of A signal, and c signal is flat. Frequency response curve. In order to make the W8 and speaker 9 can output the flat C signal, the digital signal processor 3 must first capture the A signal stored in the read-only memory 5 in order to obtain The negative B signal parameter of the a signal, and the negative B signal parameter is transmitted back to cpu2, converted back to the digital serial signal 22, and then transmitted to the digital / analog signal converter 4. In this way, the digital signal processor can be reduced. 3 The burden when doing complex calculations to avoid lowering the transmission speed, resulting in a relatively low sound quality; Digital / analog 彳 s number converter 4, which receives the negative number b L from cpU2 and converts the received 佗 number After the analog signal 41, it is sent to the filter buffer 6 for filtering and buffering, and then to the power amplifier 7 for power amplification. After that, it transmits the amplified negative B signal to _ ^ 8 and speaker 9 , So that the A signal parameter of the frequency response curve of La < 8 and speaker 9 plus the negative B signal parameter can obtain the c signal, and the c signal will be equal to "〇", so that the Rabat 8 and speaker 9 can output a flat frequency response In addition, cpu2 also calculates the operation result of the digital 仏 processor 3, and sets the maximum output level of the digital / analog signal converter 4 to prevent cutting distortion caused by the power amplifier 7 due to excessive signal input. In addition, the analog / digital signal converter 4 acquires the stereo 1244269 audio signal data obtained by the CPU2. The digital signal processor 3 can convert the low pass filter (L0w PASS FILTER) and the high pass filter (HIGH PASS FILTER). The left and right channels are added and divided by two. At the same time, a low-pass filter is added. After this signal is calculated with another speaker parameter, it is converted by another digital / analog signal. You can get a good curve of another channel (subwoofer). Furthermore, the original frequency response curve parameters of the Rabat 8 and the speaker 9 are stored in the 凟 凟 memory 5 or related similar components. Once the frequency response curve of w 8 and the speaker 9 is changed, the changed frequency The response curve parameters can be stored in the read-only memory 5 or similar devices at any time to improve the lack of conventional compensation circuits. The digital audio optimizer provided by the present invention has the following advantages when compared with other conventional technologies: 1. The present invention can at any time first properly simulate the frequency response curve of the obtained pap and speaker on the computer After adjustment, save this curve data to get the best frequency response curve. 2. In the manufacturing process of the present invention, the purpose of changing the design at any time can be accomplished by simply inserting the read-only memory storing the Ra < and the frequency response curve of the speaker onto the circuit board. ° ° 3 · The present invention converts the DSp operation result into a serial signal through a cpu, and then sends it to a digital / analog signal converter to convert it into an analog signal, in order to reduce the burden required for "^^^" when performing complex calculations. To achieve the purpose of maintaining sound quality. 4. The present invention first sets the maximum output level of the digital / analog signal converter to prevent the power amplifier from cutting distortion due to excessive signal input, so that it is not necessary to use most components and composite lines to form a compression as in the traditional way Lines to improve cutting distortion. The above detailed description is specific to one of the feasible embodiments of the present invention. The 1244269 column is not intended to limit the patent scope of the present invention. Any equivalent implementation or change that does not depart from the spirit of X Α 技 # should be included in this case. Patent scope. / As mentioned in the above, this case is not only innovative in terms of technical concept, but also can enhance the above-mentioned multiple effects compared with the conventional one. It should have fully met the requirements of the issued patents for novelty and advancement, and filed an application in accordance with the law. The Qinggui Bureau approves this invention patent application to encourage inventions to the utmost. [Brief description of the drawings] π Please refer to the following detailed description of a preferred embodiment of the present invention and the accompanying drawings to further understand the technical content of the present invention and its purpose and effect; the drawings related to this embodiment are: Figure 1 This is a block diagram of the digital audio optimizer of the present invention. [Representative symbols of main parts] 1 Analog / digital signal converter 10 Audio analog signal 11 Digital serial signal 2 CPU 21 Parallel signal 22 Digital serial signal 3 Digital signal processor 4 Digital / analog signal converter 41 Analog signal 5 Read-only Memory 6 Filter / Buffer 7 Power Amplifier 8 Port 8