TW501376B - Decoding device and method of digital audio - Google Patents
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501376 五、發明說明Ο) 【發明領域】 本發明係關於一種數位音訊之處理,特別是關於 MP3(MPEG Audio Layer 3)之解碼裝置及方法。 【發明背景】 隨著數位音樂的風行,使得數位壓縮技術也隨之蓬勃 發展。其中,數位音訊由於具有高壓縮及高音質之特性, 而廣受歡迎,亦使得許多相關之解壓縮軟硬體紛紛出籠。 第一圖為習知MP3解壓縮之方塊圖,在步驟S1 0中,數位音 訊位元流(MP3 Bit Stream)經過同步與錯誤檢查 (Synchronizing & Error checking)解出MP3 之相關資 料,如霍夫曼解碼(H u f f m a n D e c 〇 d i n g )和比例資訊 (Scalefactor Information),並檢查資料是否無誤,依 據霍夫曼表格(Huffman Table)進行霍夫曼解碼,以解出 相對應的數值,即步驟S1 2,接著進行步驟S 1 4,即反量化 (D e - q u a n t i z a t i ο η ),其係依據比例資訊將其接收之資料 加以處理,得到介於-1及+ 1之間的小數;然後再將資料重 新排列(R e 〇 r d e r ),即步驟S 1 6。 在步驟S20中,利用聯合立體聲解碼器(Joint Stereo Decoding)將信號轉成立體聲信號,分成左聲道和右聲 道;二路徑分別經過抗失真(Anti-Aliasing)處理,即步 驟S22 ;然後再進行步驟S24,即修正式反餘弦轉換 (Inverse Modified Discrete Cosine Transfer)與重疊 (Overlap);最後利用合成多相濾波集合(Synthesis501376 V. Description of the invention 0) [Field of the invention] The present invention relates to a digital audio processing, in particular to a decoding device and method for MP3 (MPEG Audio Layer 3). [Background of the Invention] With the popularity of digital music, digital compression technology has also flourished. Among them, digital audio is popular due to its high compression and high sound quality, and many related decompression software and hardware have come out. The first picture is a block diagram of the conventional MP3 decompression. In step S10, the digital audio bit stream (MP3 Bit Stream) is subjected to synchronization and error checking (Synchronizing & Error checking) to resolve MP3 related data, such as Huo Huffman Decoding and Scalefactor Information, and check whether the data is correct, and perform Huffman decoding according to the Huffman Table to solve the corresponding value, that is, steps S1 2, and then proceed to step S1 4, which is inverse quantization (D e-quantizati ο η), which processes the data received according to the proportional information to obtain a decimal between -1 and + 1; then The data are rearranged (R e 〇rder), that is, step S 1 6. In step S20, a joint stereo decoder (Joint Stereo Decoding) is used to convert the signal into a stereo signal, and is divided into a left channel and a right channel; the two paths are respectively subjected to Anti-Aliasing processing, that is, step S22; and then Step S24 is performed, that is, Inverse Modified Discrete Cosine Transfer and Overlap; finally, a synthetic polyphase filter set (Synthesis
501376 五、發明說明(2)501376 V. Description of the invention (2)
Polyphase Filterbank)將頻域(frequency domain)信號 還原成時域(t i m e d o m a i n )信號,並且加以合成濾波而得 到原來的脈碼調變(Pulse Coded Modulation ; PCM)信 號。 在第一圖所示之解碼過程中,需要大量的暫存器及運 算處理之時間,主要是由進行聯合立體聲解碼S2 0需暫存 二個聲道之資料並加以處理,以及抗失真處理S2 2亦需暫 存二個相鄰副頻帶(s u b b a n d )之資料並力tr以處理。 在信號處理的過程中,立體聲之解碼公式如下:Polyphase Filterbank) restores the frequency domain (frequency domain) signal to the time domain (t i m e d o m a i n) signal, and synthesizes and filters it to obtain the original Pulse Coded Modulation (PCM) signal. In the decoding process shown in the first figure, a large number of registers and processing time are required, mainly for joint stereo decoding S2. 0 Two channels of data need to be temporarily stored and processed, and anti-distortion processing S2 is required. 2 It is also necessary to temporarily store the data of two adjacent subbands and to process them. In the process of signal processing, the stereo decoding formula is as follows:
MS[L] [Sb][SS]=(XY[L][Sb][SS] + XY[R][Sb][SS]/ /"2 (1) MS[R] [Sb][SS]=(XY[L][Sb][SS 卜XY[R][Sb][SS]/ (2) 其中,MS為立體聲之解碼的輸出;L為左聲道;R為右 聲道;Sb為副頻帶索引(Subband index) ; SS為比例頻帶 索引(Scalefactor band index) ;Xr 為輸入信號。 而抗失真之公式如下:MS [L] [Sb] [SS] = (XY [L] [Sb] [SS] + XY [R] [Sb] [SS] / / " 2 (1) MS [R] [Sb] [SS ] = (XY [L] [Sb] [SS BU XY [R] [Sb] [SS] / (2) where MS is the output of stereo decoding; L is the left channel; R is the right channel; Sb Is the subband index; SS is the scale factor band index; Xr is the input signal. The anti-distortion formula is as follows:
An[L/R][Sb][17-SS]-MS[L/R][Sb][17-SS]XCs[SS]- MS[L/R][Sb+1 ] [SS] XCa[SS] . (3)An [L / R] [Sb] [17-SS] -MS [L / R] [Sb] [17-SS] XCs [SS]-MS [L / R] [Sb + 1] [SS] XCa [ SS]. (3)
An[L/R][Sb+l][SS]-MS[L/R][Sb+l][SS] XCs[SS]+An [L / R] [Sb + l] [SS] -MS [L / R] [Sb + l] [SS] XCs [SS] +
MS[L/R][Sb][17-SS]XCa[SS] (4) 其中,An為抗失真處理步驟S22之輸出;L/R表示左聲 道或右聲道;SS為0〜7之整數;Cs及Ca為介於(_1,+1)之間 的常數。MS [L / R] [Sb] [17-SS] XCa [SS] (4) Among them, An is the output of anti-distortion processing step S22; L / R means left channel or right channel; SS is 0 ~ 7 Integer; Cs and Ca are constants between (_1, +1).
第5頁 ^01376 五、發明說明(3) 對應前述之公式,一 中所示。其中包括左聲St,,聲解碼器之袭置如第二圖 衝級暫存器B3等三個鲂六,存^B1、右聲道暫存器B2、緩 576字元(word)。左 子為,在每一個暫存器的容量為 提供左聲道與右聲道^管士存器β 1及右聲道暫存器B2分別 Β 3則用來接收下一敕^^日$所需之記憶體,緩衝級暫存器 組加法器1 〇、一組二==輸入信號。而此架構運算時需一 解碼參數,例如1 / 器1 2、二組乘法器1 4與1 5及立體聲 塊 第 元 另 【ii顯ϊΐ:,中進行抗失真處理s22之硬體方 副頻帶暫存哭=等存器β4、第二副頻帶暫存器M以及 其中兩個暫;哭個記憶體,其可儲存3X18個字 個暫在哭目,丨Γ:為個副頻帶運算時所需之記憶體, 其運算過程;接,:-個副頻帶的輸入信號。而在 一 & π I /、而—組乘法器20與22、一組加法器24、 ,'^抗失真參數,例如Ca[SS]與Cs[SS]。 严舻丘9 yIt!知ύ立體聲解碼和抗失真處理所需之暫存記 ϊ Λ „3Χ1 8 = 1 78 2字元,其運算時需要二組加法 =日了 i'法15及四組乘法器,這佔整個解碼系統相當大 <日日片面積。 蓺齡因你此立1有2監於上述習知技藝之缺失,一種優於習知技 ☆數位曰成的解碼裝置與方法乃為所冀。 【發明目的與概述】 本發明的主要目的係在於提出一種將數位音訊解瑪器Page 5 ^ 01376 V. Description of the invention (3) Corresponding to the aforementioned formula, one is shown in. This includes the left sound St. The sound decoder is set as shown in the second figure, the three-level register B3, etc., and stores ^ B1, the right channel register B2, and 576 characters. The left sub is to provide the left channel and the right channel ^ in the capacity of each register. The tube register β 1 and the right channel register B 2 respectively B 3 are used to receive the next ^^ day $ The required memory is a buffer-level register group adder 10 and a group of two == input signals. And this architecture requires a decoding parameter for calculation, such as 1/2, two sets of multipliers 1 4 and 15, and the stereo block element [ii show: in the hardware side sub-band for anti-distortion processing s22 Temporary cry = equal register β4, the second sub-band register M, and two of them; a cry memory, which can store 3 × 18 words temporarily crying, Γ: for the sub-band operation Required memory, its calculation process; then:-an input signal of the sub-band. And in a & π I /, and a set of multipliers 20 and 22, a set of adders 24, and ^ anti-distortion parameters, such as Ca [SS] and Cs [SS]. Yan Yanqiu 9 yIt! Known temporary storage needed for stereo decoding and anti-distortion processing ϊ Λ „3 × 1 8 = 1 78 2 characters, the operation requires two sets of addition = day i 'method 15 and four sets of multiplication Device, which occupies a large area of the entire decoding system, <day-to-day and film area. Since you are here, you have 2 and 2 due to the lack of the above-mentioned conventional techniques, a decoding device and method that is superior to conventional techniques. [Objective and Summary of the Invention] The main object of the present invention is to propose a digital audio demapping device.
501376 五、發明說明(4) 之聯合立體聲解碼及抗失真之運算整合之裝置及方法,以 減少所佔整個系統的晶片面積,並提升信號處理的速度及 效率。 根據本發明,一種數位音訊之解碼裝置與方法,係將 立體聲解碼參數及抗失真參數先行合併,再將立體聲解碼 與抗失真解碼之處理順序調換,使得每一聯合立體聲解碼 之輸出可減少一個乘法運算,對於整體資料運算處理之速 度可提昇近一倍左右,且其所需暫存記體之容量可以減少 一半以上。 【詳細說明】 由於立體聲解碼與抗失真處理之處理先後順序並不會 衫響數位音訊解壓縮之輸出結果,因此,將上述所提到之 立體聲解碼及抗失真處理的運算公式之立體聲解碼參數及 抗失真參數先行合併處理,即得到新的兩組參數, nCs[SS]:l/XCs[SS] (5) nCa[SS] = l/ ^/~ 2)x(Ca[SS] ( 6 ) 將立體聲解碼及抗失真處理的順序調換,調換後得到 抗失真的公式如下:501376 V. Invention description (4) The device and method for combining stereo decoding and anti-distortion operation integration to reduce the chip area occupied by the entire system and improve the speed and efficiency of signal processing. According to the present invention, a digital audio decoding device and method combine a stereo decoding parameter and an anti-distortion parameter first, and then switch the processing order of the stereo decoding and the anti-distortion decoding so that the output of each joint stereo decoding can be reduced by one multiplication Computing, the overall data processing speed can be nearly doubled, and its temporary memory capacity can be reduced by more than half. [Detailed description] Because the processing order of stereo decoding and anti-distortion processing will not affect the output result of digital audio decompression, therefore, the stereo decoding parameters of the above-mentioned calculation formulas of stereo decoding and anti-distortion processing and The anti-distortion parameters are merged first to obtain two new parameters, nCs [SS]: l / XCs [SS] (5) nCa [SS] = l / ^ / ~ 2) x (Ca [SS] (6) The order of stereo decoding and anti-distortion processing is reversed, and the formula for anti-distortion after the exchange is as follows:
An[L/S][Sb][SS]=Xr[L/S][Sb][17-SS] XnCs[SS]-An [L / S] [Sb] [SS] = Xr [L / S] [Sb] [17-SS] XnCs [SS]-
Xr[L/S][Sb + 1 ] [SS] XnCa[SS] (7)Xr [L / S] [Sb + 1] [SS] XnCa [SS] (7)
An[L/S][Sb + 1][SS]=Xr[L/S][Sb+1 ] [SS] XnCs[SS] +An [L / S] [Sb + 1] [SS] = Xr [L / S] [Sb + 1] [SS] XnCs [SS] +
Xr[L/S][Sb][ 17-SS] XnCa[SS]Xr [L / S] [Sb] [17-SS] XnCa [SS]
501376 五、發明說明(5)501376 V. Description of the invention (5)
Sb二0〜31 , SS=0〜7 經整理後得到之運算式為:Sb two 0 ~ 31, SS = 0 ~ 7 After finishing the calculation formula is:
An[L/S][Sb][others]=Xr[L/S][Sb][others]/y^ 2 (9) 其中,當Sb為0時,others為SS = 0〜7 當Sb 為 1 〜30,others 為SS=8,9 〇 當Sb二31 ,others 為SS=8〜17 〇 而聯合立體聲之運算公式則轉換成: MS[L][Sb][SS]=An[L][Sb][SS]+An[R][Sb][SS] (10) MS[R][Sb][SS]=An[L][Sb][SS]-An[R][Sb][SS] (11) 與背景所述之第(1)式及第(2)式比較,每一MS立體聲解碼 之輸出可減少一個乘法運算,而一個聲道有3 2 X1 8個輸 出。換言之,本發明所提出的方法在處理一個聲道的資料 過程可較過去省去576個乘法運算,而提昇整體之運算處 理速度。 第四圖顯示本發明之一實施例,說明上述運算方法之 硬體方塊。首先,輸入信號Xr,經過乘法器3〇與參數提供 者34提供之參數,例如1/、nCa[SS]或nCs[SS],此來、 數係由一組計數器3 2所選擇,若輸入信號Xr所乘之參數'為 1 / /~2,則所得之結果直接儲存至暫存器3 5 ;否則每>一個… X 則乘以與其對應的n c a [ s s ]及n C s [ S S ],所得到之結果分 別儲存至暫存器3 6及3 7,直到一組副頻帶之資料處^完刀An [L / S] [Sb] [others] = Xr [L / S] [Sb] [others] / y ^ 2 (9) Where, when Sb is 0, others is SS = 0 ~ 7 When Sb is 1 to 30, others is SS = 8, 9 〇 When Sb is 31, others is SS = 8 to 17 〇 The joint stereo formula is converted to: MS [L] [Sb] [SS] = An [L] [Sb] [SS] + An [R] [Sb] [SS] (10) MS [R] [Sb] [SS] = An [L] [Sb] [SS] -An [R] [Sb] [ SS] (11) Compared with formulas (1) and (2) described in the background, the output of each MS stereo decoding can be reduced by one multiplication operation, and one channel has 3 2 X 1 8 outputs. In other words, the method proposed by the present invention can save 576 multiplication operations in the process of processing the data of one channel, and improves the overall processing speed. The fourth figure shows an embodiment of the present invention, illustrating a hardware block of the above-mentioned calculation method. First, the input signal Xr passes through the parameters provided by the multiplier 30 and the parameter provider 34, such as 1 /, nCa [SS] or nCs [SS]. Here, the number system is selected by a set of counters 32. The parameter 'multiplied by the signal Xr is 1 / / ~ 2, and the obtained result is directly stored in the register 3 5; otherwise, each > X is multiplied by the corresponding nca [ss] and n C s [SS ], The obtained results are stored in the temporary registers 36 and 37 respectively, until the data of a group of sub-bands is finished.
501376 五、發明說明(6) 成。此時,將暫存器3 6及3 7之資料讀出,與下一組副頻帶 之輸入信號完成乘法運算後之結果,經過加/減法器3 8進 行相加或相減。在此過程中,同一時間只完成一個動作。 重複以上動作直到暫存器3 5填滿,即一個聲道的資料 32 XI 8字元。當一個聲道之資料處理完成,若下一個聲道 的資料完成處理後,即可與暫存器3 5所對應的資料讀出, 並同時經過加法器4 0及減法器4 2進行加法或減法運算。如 此,即可同時得到左、右兩聲道之資料輸出。 由第四圖中所示,所使用之運算單元包括一組乘法器 3 0、一組加減法共用之加/減法器3 8、一組加法器4 0及一 組減法器4 2。相較於第二圖所示之習知技藝中聯合立體聲 與抗失真處理,本發明在硬體上節省了三組乘法器。在暫 存器方面,第四圖中所示之電路所需之記憶體包括暫存器 36(8字元)、暫存器37(8字元)、暫存器35(32X18)加上暫 存器3 9 ( 8字元),總共需6 0 0字元之記憶體,相較於第三圖 所示之習知技藝所需1 7 8 2字元,明顯大幅度地減少。 因此,本發明k將立體聲解碼參數及抗失真參數先行合 併處理,再將立體聲解碼與抗失真解碼之處理順序調換, 使得每一聯合立體聲解碼之輸出可減少一個乘法運算,進 而提昇整體之運算速度、除此之外,利用本發明所提出之 裝置及方法,'更使得所需之記憶體減少,進而節省硬體的 空間'並且降低成本。再者,若將第四圖中所示之架構搭配 管線式(pipeline)的處理方法,如第五圖中所示,在每一 個周期中,各運算單元及暫存器持續的對輸入進來之數位501376 Fifth, the description of the invention (6). At this time, the data of the registers 36 and 37 are read out, and the result of the multiplication with the input signal of the next set of sub-bands is added and subtracted by the adder / subtractor 38. In this process, only one action is completed at a time. Repeat the above operation until the register 3 5 is filled, that is, one channel of data 32 XI 8 characters. When the processing of the data of one channel is completed, if the processing of the data of the next channel is completed, the data corresponding to the register 35 can be read out, and at the same time, the adder 40 and the subtractor 4 2 can be added or Subtraction. In this way, the data output of the left and right channels can be obtained at the same time. As shown in the fourth figure, the operation unit used includes a set of multipliers 30, a set of adder / subtractors 38 common to addition and subtraction, a set of adders 40 and a set of subtractors 42. Compared with the combination of stereo and anti-distortion processing in the conventional technique shown in the second figure, the present invention saves three sets of multipliers on the hardware. Regarding registers, the memory required for the circuit shown in the fourth figure includes register 36 (8 characters), register 37 (8 characters), register 35 (32X18) plus temporary The memory of 39 (8 characters) requires a total of 600 characters of memory, which is significantly reduced compared to the 172 characters required for the conventional art shown in the third figure. Therefore, in the present invention, the stereo decoding parameters and the anti-distortion parameters are combined and processed first, and then the processing order of the stereo decoding and the anti-distortion decoding is switched, so that the output of each joint stereo decoding can be reduced by one multiplication operation, thereby improving the overall operation speed. In addition, using the device and method proposed by the present invention, 'reduced the required memory, thereby saving hardware space' and reducing costs. Furthermore, if the architecture shown in the fourth figure is matched with a pipeline processing method, as shown in the fifth figure, in each cycle, each computing unit and register continuously inputs the input. digit
501376 五、發明說明(7) 音訊資料進行運算處理,則數位音訊解壓縮之速度將可以 再提昇。 以上對於本發明之較佳實施例所作的敘述係為闡明之 目的,而無意限定本發明精確地為所揭露的形式,基於以 上的教導或從本發明的實施例學習而作修改或變化是可能 的,實施例係為解說本發明的原理以及讓熟習該項技術者 以各種實施例利用本發明在實際應用上而選擇及敘述,本 發明的技術思想企圖由以下的申請專利範圍及其均等來決 定0501376 V. Description of the invention (7) The audio data is processed and processed, and the speed of digital audio decompression can be further improved. The above description of the preferred embodiment of the present invention is for the purpose of clarification, and is not intended to limit the present invention to exactly the disclosed form. Modifications or changes are possible based on the above teaching or learning from the embodiments of the present invention. The embodiments are selected and described in order to explain the principle of the present invention and allow those skilled in the art to use the present invention in practical applications in various embodiments. The technical idea of the present invention is intended to be based on the scope of the following patent applications and their equivalents. Decision 0
第10頁 501376 圖式簡单說明 對於熟習本技藝之人士而言,從以下所作的詳細敘述 配合伴隨的圖式,本發明將能夠更清楚地被瞭解,其上述 及其他目的及優點將會變得更明顯,其中: 第一圖係習知之MP3解壓縮之方塊圖; 第二圖係第一圖中聯合立體聲解碼器之方塊圖; 第三圖係第一圖中抗失真處理之方塊圖; 第四圖係本發明之一實施例;及 第五圖係第四圖中之配合管線式信號處理的示意圖。 圖號說明: 1 0,2 4,4 0,4 2 加法器 1 2,2 6減法器 1 4,1 5,2 0,2 2,3 0 乘法器 3 2計數器 3 4參數提供者 35, 36,37, 39 暫存器 3 8加/減法器501376 Schematic description for those skilled in the art, the present invention will be more clearly understood from the detailed descriptions and accompanying drawings made below, and its above and other objectives and advantages will change. It is more obvious, in which: the first picture is a conventional block diagram of MP3 decompression; the second picture is a block diagram of a joint stereo decoder in the first picture; the third picture is a block diagram of anti-distortion processing in the first picture; The fourth diagram is an embodiment of the present invention; and the fifth diagram is a schematic diagram of the pipeline signal processing in the fourth diagram. Drawing number description: 1 0, 2 4, 4 0, 4 2 Adder 1 2, 2 6 Subtractor 1 4, 1 5, 2 0, 2 2, 3 0 Multiplier 3 2 Counter 3 4 Parameter Provider 35, 36, 37, 39 Register 3 8 Adder / Subtractor
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Cited By (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US8756066B2 (en) | 2007-02-14 | 2014-06-17 | Lg Electronics Inc. | Methods and apparatuses for encoding and decoding object-based audio signals |
TWI476762B (en) * | 2010-08-13 | 2015-03-11 | Ntt Docomo Inc | Audio decoding device, audio decoding method, audio decoding program, audio coding device, audio coding method, and audio coding program |
US9609279B2 (en) | 2004-09-24 | 2017-03-28 | Google Inc. | Method and system for providing secure CODECS |
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2001
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Cited By (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US9609279B2 (en) | 2004-09-24 | 2017-03-28 | Google Inc. | Method and system for providing secure CODECS |
US10691778B2 (en) | 2004-09-24 | 2020-06-23 | Google Llc | Method and system for providing secure codecs |
US8756066B2 (en) | 2007-02-14 | 2014-06-17 | Lg Electronics Inc. | Methods and apparatuses for encoding and decoding object-based audio signals |
US9449601B2 (en) | 2007-02-14 | 2016-09-20 | Lg Electronics Inc. | Methods and apparatuses for encoding and decoding object-based audio signals |
TWI476762B (en) * | 2010-08-13 | 2015-03-11 | Ntt Docomo Inc | Audio decoding device, audio decoding method, audio decoding program, audio coding device, audio coding method, and audio coding program |
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