TW293229B - Filter network reverberation generator - Google Patents

Filter network reverberation generator Download PDF

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Publication number
TW293229B
TW293229B TW85105898A TW85105898A TW293229B TW 293229 B TW293229 B TW 293229B TW 85105898 A TW85105898 A TW 85105898A TW 85105898 A TW85105898 A TW 85105898A TW 293229 B TW293229 B TW 293229B
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Taiwan
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filter
wave
item
reverb
patent application
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TW85105898A
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Chinese (zh)
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Wenn-Yuh Su
Lii-Woei Wang
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Ind Tech Res Inst
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Abstract

A filter network reverberation generator for processing the audio signals, and which includes: 1. a plurality of wave-ladder filters, each of them simulated the audio signals sent or reflected by wave transmission from a certain direction,input the original audio signals into the wave-ladder filter and created different oscillation in each direction; 2. At least one of the wave-ladder filter will contain one or several secondary space distribution matrix to simulate the middle reflex object; 3. An adder which combined with the output signals of all the wave-ladder filters to create the reverberation signals.

Description

經濟部中央標準局員工消費合作社印製 ^3229 Λ7 Β7 五、發明説明(I ) 發明背景 這篇發明是關於聲音訊號的殘響產生器,殘響 (reverberation)通常產生在一具有反射物體的聲學空間 內’它產生的原因是由原始波、伴隨發生由地板、牆壁及 其他物體的反射波合起來的多重反射。一般而言,在一聽 起來悅耳的聲學空間內都具有像指數函數(exponential function)衰減般的能量樣本。既然人類的聽覺系統是比 較習慣於處在有殘響的環境內,所以在一無殘響環境內的 錄音通常聽起來也較不自然了。 時至今日,許多在工作室的錄音都加入一些阻尼作用 (damping effect),以致於殘響效果幾乎不見了。爲了要 讓錄音結果聽起來像是在眞實的空間(如體育場)所產生 的,就使用一些後處理的方式,用人工所產生的殘響是其 中最普遍的。 用人工殘響(artificial reverberation)裝置必須要產 生原來的信號加上一自然在聲學環境中產生的殘響。一個 聽起來悅耳的殘響裝置至少必須滿足下列的準則,聲音能 量的樣式必須像指數函數般的衰減’在聲音頻譜中頻部份 通常比低頻及高頻含有較長的殘響。而各方向的反射以強 度與時間長短而言必須是不規則的。而此系統必須依上述 法則處理早期反射(early reflection)與長殘響(l〇ng reverberation) ° 本紙乐尺度適用中圈困家標準(CNS ) A4規格(210 XW7公釐) (請先閲讀背面之注意事項再填寫本頁) •-°Printed by the Employee Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs ^ 3229 Λ7 Β7 V. Description of the invention (I) Background of the invention This invention relates to a reverb generator of sound signals. Reverberation usually produces a sound with a reflective object The reason for it in the space is that it is caused by the original wave and the multiple reflections combined by the reflected waves from the floor, walls and other objects. Generally speaking, in an acoustic space that sounds pleasant, there are energy samples that decay like an exponential function. Since the human auditory system is more accustomed to being in a reverberant environment, recordings in a reverberant environment usually also sound less natural. Today, many recordings in the studio have added some damping effect (damping effect), so that the reverb effect is almost gone. In order to make the recording result sound like it is produced in a real space (such as a stadium), some post-processing methods are used, and the artificial reverberation is the most common among them. Artificial reverberation (artificial reverberation) devices must produce the original signal plus a natural reverberation in the acoustic environment. A pleasant reverb device must meet at least the following criteria. The sound energy pattern must be attenuated exponentially. The mid-frequency portion of the sound spectrum usually contains longer reverb than the low and high frequencies. The reflection in all directions must be irregular in terms of intensity and length of time. This system must deal with early reflection and long reverberation according to the above-mentioned rules. This paper music scale is applicable to the A4 specification (210 XW7 mm) of the Central Circle Standard (CNS) (Please read the back first Note to fill out this page) •-°

T 經濟部中央標準局貝工消费合作社印製 A7 B7 五、發明説明(>) 傳統的人工殘響技術是使用機械裝置或利用電子電路 模擬機械裝置,但隨著數位信號處理(Digital Signal Processing)技術的快速發展,現代的人工殘響裝置就大 多是數位的。在眾多良好的處理方法中,有一種產生悅耳 殘響是測量一個良好聲學空間的脈衝反應(irnpulse response)然後使用這反應產生有限脈衝響應(Finite Impulse Response(FIR))與原始信號做迴旋運算 (convolution)產生輸出。通常濾波器的節數(taps)少至 幾仟個,多至數萬個,這種運算量是非常可觀的。然而在 大多數的應用中,這種方式的實踐是非常不實際的,因 此’太多數的數位殘響產生器是結合兩種方式,一爲早期 反射(early reflection)是使用有限脈衝響應,而長殘響 是使用無限脈衝響應(Infinite Impulse Response(IIR)) 來表示’以便更進一步降低運算量。然而,在殘響器中使 用無限脈衝響應濾波器(IIR filter)時,有些困難便因此 產生了。首先,它產生一種令人耳覺得煩擾的梳型濾波器 效應(comb filter effect)。再來,在接近殘響要結束 時’聲音容易變成較亮,而有時較刺耳。第三,若有任何 瀘波器是位於無限脈衝響應濾波器(IIR)的回授迴圈 (feedback loop)時,穩定性的問題便是一個難以處理的 課題了。最後,要將無限脈衝響應濾波器的特性去模擬聲 音的特性時,亦是另一花工夫的課題。 發明摘要 傳統方式之所以難以產生高品質的殘響的原因之一’ 是這些技術基本的模型根本就沒有將聲學環境的因素考慮 本紙張尺度適用中國國家標準(CNS ) A4規格(210X297公釐) (請先閲讀背面之注意事項再填寫本頁) 、-0 A7 B7 293229 五、發明説明(汐) 進去,因此,上述之人工殘響(artificial reverberation:) 方式才由一眞實的空間激發以產生殘響。 因此,我們建立一個以模擬虛設聲學空間(imaginary acoustic room)爲基礎的殘響技術。一般而言,這技術是 以一種數位波階梯濾波器(wave-ladder filter)爲基礎, 去模擬一維聲波的傳播。一虛擬空間是根據這空間的幾何 分佈,由好幾個數位波階梯濾波器,並模擬在各個位置點 上,聲波交會時各方向量重新分配的情形。數位波階梯濾 波器加上空間分配矩陣(Spatial Distribution ]^311^^(801^))就'形成了一人工殘響產生器,而這殘響 產生器的穩定性亦可容易達成,只要這些濾波器與空間分 配矩陣從能量的觀點而言是會損失的(lossy)。這系統在 不影響穩定性的前提下也非常容易調整系統的參數,如這 虛擬空間的大小及殘響的增益。爲了產生眞實的早期反射 (early reflection),在信號進入系統濾波網路前,我們 使用了有限脈衝響應(FIR)濾波器來模擬。’ 通常而言,在這篇發明的槪念中,它只是濾波器網路 的集合,濾波器網路的基本元件是由一維波形傳播的模型 所推導而得的,而空間分配矩陣(SDM)用來連接各個基 本元件。而這網路的結構是用來模擬一個虛擬的聲學空間 中聲波的傳播行徑。而聲音信號輸入有限脈衝響應濾波器 (FIR)獲得早期反射(early reflection),結果再輸入這濾 波器網路獲得長殘響(long reverberation)後’殘響的效 果便產生了,而這濾波器網路的穩定性亦可得而證之。 本紙張尺度適用中國國家標準 (CNS ) Α4ϋ^( 210X297公釐) (請先閲讀背面之注意事項再填寫本頁) 裝' 訂 經濟部中央標準局負工消費合作社印製 2932^9 A7 B7 五、發明説明(屮) 一般而言,這篇發明的每一個元件,對於在數位信號 處理與空間音響學領域的人而言,都是已知的。虛擬空間 (imaginary room)、階梯濾波器(ladder filter)與空間分 配矩陣(Spatial Distribution Matrices)對於專精於此領 域者亦爲已知。所以,這篇發明是結合這些已知的技術及 技巧而獲得好的殘響(如聽起來悅耳、穩定、有彈性等的 好處)。從另一個角度看本發明爲,只要使用波傳播的原 理就可以輕易的產生聽起來悅耳的殘響產生器。 本發明可以產生一聽起來比傳統方式更自然、眞實的 人工殘響,它可以應用於聲音信號的處理,亦可應用在家 庭與商業環境的設定上,如劇院、音樂廳、卡拉0K等。 另外的好處與特性將因爲下列提出更具體的描述與申 請專利的範圍而更加淸楚。 圖例的簡要描述 圖一是一產生一維波傳播離散時間(discrete-time)系統 的方塊圖。 經濟部中央標準局貝工消费合作社印製 I 产 訂 (請先閱讀背面之注意事項鼻填寫本頁) 圖二是一模擬在一交接點上,信號反射與傳送的方塊 圖。 圖三是一離散時間方式實踐,有多重反射交接點的一維 波傳導的方塊圖。 圖四是一以離散時間方式實踐,有多重反射交接點的一 維波傳導,加上使用衰減器(attenuator)來模擬一 衰減的(lossy)信號路徑的方塊圖。 本紙張尺度遑用中國國家樣準(CNS ) Α4規格(21 Οχ 297公兼) A7 B7 _ 五、發明説明(5) 圖五是一以離散時間方式實踐,有多重反射交接點的τ 維波傳導,並且使用低通濾波器(low-pass filter) 於信號路徑中,以模擬一衰減的信號路徑的方塊 圖。 圖六是以一有限脈衝響應(FIR)濾波器來模擬’早期反 射的方塊圖。 圖七是以一無限脈衝響應(IIR)實踐的全通濾波器(all_ pass filter)的方塊圖。 圖八是以一無限脈衝響應(IIR)實踐的梳形滴波器 (comb filter)的方塊圖。 圖九是一使用有限脈衝響應(FIR)濾波器’後面接四個 波階梯濾波器的濾波器網路殘響產生器的方塊圖° 圖十是一數位濾波器網路殘響產生器及它旁所支援的功 能,而這正是圖ll.b之實踐。 圖Π.a與圖1 l.b是濾波器網路殘響產生器所能建立的虛 擬空間的範例。 ’ --------{裝-- (請先聞讀背面之注意事碩再填寫本頁) ,vs i; 經濟部中央標隼局員工消費合作社印製 發明的具體描述 本發明的實施例是用作音訊錄音後處理的殘響產生 器。輸入的聲音訊號分成二階段處理’有限脈衝響應 (FIR)濾波器模擬早期反射’然後一濾波器網路模擬長殘 響。殘響產生器是以模擬聲波傳播爲基礎的’也就是說’ 將連續時間(c ο n t i n U 〇 u s -1 i m e)—維波傳導轉換成可以以 圖一之數位濾波器10實踐的離散時間模型。以數位形式 模擬波傳導對在數位信號處理這領域的人來說是已知的’ - 6 - 本紙張尺度適用中國國家揉準(CNS ) A4规格(210X297公簸Ί A7 B7 29S22d 五、發明説明(6 ) 而且有多本書關於這方面的可以做爲參考’如L.R. Rabiner et al., Digital Processing of Speech Signals, Prentice-Hall, Inc·, 1978。 圖一展示出一個模擬一維波傳導無損失(lossless)的 數位波階梯濾波器’它模擬直接由一個物體’如牆壁’反 射回來。輸入是原始音訊,輸出訊號爲經過反射後返回原 來信號位置點的聲音,這簡單數位波階梯濾波器包括模擬 上分路(upper branch)之方向向外的聲波,及下分路 (lower branch)方向爲從牆壁反射回來之聲波。每一分路 是由幾組延遲模組16 (delay module)所建立的,在每— 分路中模組的數目及每一模組中延遲的量,都由我們所要 模擬的虛擬聲學空間的大小來決定。一增益模組18 (gain module)連接從上分路12至下分路Μ後,它相當於聲音從 牆壁反射回來的量’通常反射量的大小會小於1,所以增 益模組之增益亦會小1。增益模組中的增益及延遲模組^ 的延遲都是可程式化的(programmable),因此,使用者 可以由此調整各個參數以便符合聲音的特性。 既然聲波可縱雜端到最後反始_細移動,這 情形可以由圖二與圖三中模擬出來。圖二顯示出—無損失 的數位波階梯其中的-段’它由空_分配矩陣24 (s 作爲模型模擬-個-維波_神間反娜,雖然圖上沒 有__ ’但是遠_牆_似以1益模組連接讀 路右_的上分_下分路來代表之’正細前面圖一所 述。在此範例中,使用了四組延遲模組20(1),2 ), 本紙张尺度遙用中國國家橾準(CNS > A4規格(210X297公釐) (請先閲讀背面之注意事項再填寫本頁) 裝 經濟部中央橾隼局貝工消费合作社印製 經濟部中央標準局員工消費合作社印裝 A7 B7 五、發明説明(7 ) 22(1)和22(2)。延遲模組20(1)是代表聲波從一事先決定 好的位置點傳遞到由空間分配矩陣(SDM)所代表的接點 之間所花費的時間,延遲模組22(1)是代表反射的聲波從 空間分配矩陣(SDM)經過和上面所述同樣的路徑返回到 那先決定好的位置點所花費的時間。因此,模組20(1)與 22(1)延遲量的確是一樣的,同理可以推測到所有的延遲 模組’若是位於同樣聲學路徑段落上,應該有相同的延遲 圖二之網路是用來模擬一聲波通過如同一門口的交接 點,再進入一不同大小鄰接的空間,所以部份的聲波通過 交接點,而部份的聲波反射回去。同理,返回的聲波在交 接點上亦經歷同樣的過程,也就是說,部份的聲波通過交 接點回到主要的空間,而部份的聲波又返回到鄰接的空間 內。若空間內確實的情形能夠精確計算出來,那麼在每一 交接點反射與傳送出去的量便能精確的決定了。然而,這 些空間內確實的情形是相當難模擬與測量的。因此,比較 實際的作法是產生一個如前所述之虛擬的聲學空間,然後 再調整控制參數,如空間分配矩陣(SDM)的延遲量與係 數,去製造出所需要的聲學殘響效應。 圖二中之空間分配矩陣(SDM)24是以對角係數 (d i a g ο n a 1 c 〇 e f f i c i e n t s)模擬聲波通過交接點和不在對角 上的係數(〇 f f - d i a g ο n a I c 〇 e f f i c i e n t s)模擬聲波從交接點 反射回去的情形,此例中它是一個二階(2X2)的矩陣圖 五表示了三個如圖四所示的波階梯濾波器連接在一起的情 一 8 一 本紙張尺度適用中國國家標準(CNS ) A4規格(210X297公釐) ---------< .装— (请先閲讀背面之注意事項真填寫本萸) -- 經濟部中央標準局員工消费合作社印裝 A7 B7 五、發明説明(8) 形,這種組合可以用來模擬在原始接點與牆壁之間的聲學 路徑多個部分反射的物體,每一中間的物體反射回去部份 的聲波。 延遲的量由所要模擬的虛擬空間大小所決定,埋論上 而言,反射係數亦可由虛擬空間的特性,如幾何形狀和空 氣密度,所計算而來的。然而,在實際上是有非常大的彈 性,但是也因爲如此,如何產生好的聲學結果才是主要的 選擇方向。 既然聲學能量通常在空間中傳播時會逐漸衰減,在波 階梯濾波器上加入一衰減的因素是符合實際之情況的,如 圖四及圖五。圖四與圖五都表示一個由空間分配矩陣所代 表的一維波傳導反射的情形,在交接點間的傳播路徑再加 上一有衰減(lossy)的物體。除了在聲波路徑上加入模擬 能量衰減的模組外,圖三與圖四、五是相似的。在圖四的 例子當中,模擬出發(outgoing)聲波的分支,加上了分別 帶有增益叾,+ 1與叾^2的增益模組30(1),30(2),而模擬反 射回來聲波的分支,亦加上了分別帶有增益§1£ + 1與§1£+2的 增益模組32(1)與32(2)。在圖五的例子中,加入於分支 的模組濾波器模組34(1)與34(2),在另外—分支所加入的 模組是濾波器模組36(1)與36(2)。 如圖四所示,在聲學路徑中同樣一部分的位置的增益 模組應該要有相同的增益。因此,同樣的衰減係數便要應 用在同一部分的聲學路徑,但方向是相反的增益模組上 -9 ~ 本紙張尺度適用中國國家棣準(CNS ) ( 2丨0X2.97公釐) (請先閱讀背面之注意事項再填寫本頁) .裝. 訂 ^93229 A7 B7 五、發明説明(烊) -··. 了。換言之’增益模組3〇(1)與32(1)應有相同之値,增益 模組3〇(2)與32(2)亦有相同之値。 在圖五所示之有衰減的(lossy)數位波階梯濾波器中, 濾波器可以是如圖7與8所示的有限脈衝響應(FIR)瀘波器 或是無限脈衝響應(IIR)濾波器。而這二者都可以在現行 的濾波器設計敎科書中更進一步的闡揚與探討。通常低通 濾波器(low-pass filter)代表空氣中吸收性,如眾所知, 高頻信號通常比低頻信號衰減快多了,當然了,在設計一 虛擬空間時,使用者可以不加上上述之參數。然而,模擬 低頻信號衰減較快亦無可厚非,只要聽起來好聽就好了。 在任一設計中,使用者可以使用任一濾波器去控制衰減 率。 若將上述波階梯濾波器應用在模擬早期反射(early reflection)中,通常會花上太多的記憶體奠計算量,爲了 減低計算量與記憶體,如圖六所示之有限脈衝響應(FIR) 濾波器40可以用來模擬早期反射。延遲模組42,通常每 一延遲不一定有相同的延遲量與比重(weight)G,,如圖六 所示,因此,濾波器的輸出爲輸入信號“的函數: output = Gmik m + Gm.iik.m + i +......+ Gi ik.i + G〇ik 早期反射(early reflection)的觀念在 Elements of computer Mu sic由J· Moor所著的書中都有探討。通 常’早期反射是指這些比較接近來源體由地板、牆壁或其 ____ _-10 -______ 本紙张尺度逋用中圏國家標李(CNS ) A4规格(210X297公釐) (請先閲讀背面之注意事項再填寫本頁) 裝 經濟部中央橾準局員工消費合作社印製 A7 B7 經濟部中央樣準局員工消費合作社印裝 五、發明説明() 他物體所反射回來的聲波。通常’早期反射的延遲時間大 槪分佈於0到100毫秒。 有限脈衝響應濾波器(FIR filter)中的比重與延遲量 的決定對於在信號處理這領域的人來說都是已知的’所以 細節就不再詳述。然而,通常最好的方式是直接測量如紐 約市內卡內基音樂廳之類良好聲學空間的脈衝響應’在 100毫秒內即可,量到的脈衝響應即可使用在這有限脈衝 響應(FIR)濾波器內,但這種方式將會費時又費力的。 圖九是一個由上述數位波階梯濾波器所建立而成殘響 器的方塊圖一數位化的輸入信號被有限脈衝響應(FIR)濾 波器50處理後產生早期反射(early reflection)效果。這 個被濾波後的信號再被輸入數位波階梯濾波器5 2網路 內,產生長殘響(long reverberation)。 在這個實施例中,包括四個數位波階梯濾波器52經由 空間分配矩陣54耦合。數位波階梯濾波器根據如下的原 理建立成。空間分配矩陣54結合從波階梯濾波器回饋而 來的信號(例如標記爲P丨的信號’ j爲U...,4),再產生輸 出信號(例如標記爲P丨的信號’ j爲1,...,4),再與經由有 限脈衝響應(FIR)濾波後的信號結合後,產生各別進入數 位波階梯濾波器的輸入信號。空間分配矩陣(SDM)54計 算各個由數位波階梯濾波器所模擬的信號路徑所反射回饋 回來信號的聲波壓力。然後,空間分配矩陣(SDM)54計 算輸出向外再回饋到數位波階梯濾波器的聲波壓力(例如 P:)。 -11 - --------{ 袭------訂 (請先閱讀背面之注意事項存填寫本頁) 本紙浪尺度通用中國國家榡準(CNS ) A4洗格(2丨0><297公釐) 293229 A7 B7 五、發明説明(U) 有四個分路的空間分配矩陣(SDM)54,可以以下列矩 陣運算實踐: ^T A7 B7 printed by the Beigong Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economy V. Description of the invention (>) Traditional artificial reverb technology uses mechanical devices or electronic circuits to simulate mechanical devices, but with digital signal processing (Digital Signal Processing ) With the rapid development of technology, modern artificial reverberation devices are mostly digital. Among the many good processing methods, one way to produce a pleasing reverb is to measure the impulse response (irnpulse response) of a good acoustic space and then use this response to produce a finite impulse response (Finite Impulse Response (FIR)) and the original signal to do a round-robin operation ( convolution) produces output. Usually, the number of filters (taps) is as few as a few thousand, and as many as tens of thousands. This kind of calculation is very considerable. However, in most applications, the practice of this method is very impractical, so 'too many digital reverb generators are a combination of two methods. One is early reflection (early reflection) uses a finite impulse response, and Long reverberation is expressed in Infinite Impulse Response (IIR) in order to further reduce the amount of calculation. However, when using an infinite impulse response filter (IIR filter) in the reverberator, some difficulties arise. First, it produces a comb filter effect that is annoying to the ear. Again, near the end of the reverberation, the 'sound tends to become louder, and sometimes harsher. Third, if any filter is located in the feedback loop of an infinite impulse response filter (IIR), the stability issue is an intractable issue. Finally, the characteristics of infinite impulse response filters to simulate the characteristics of sound is another problem. Summary of the Invention One of the reasons why traditional methods are difficult to produce high-quality reverberations' is that the basic models of these technologies do not consider the acoustic environment at all. This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (210X297 mm) (Please read the precautions on the back before filling out this page), -0 A7 B7 293229 V. Description of invention (Xi) Go in, therefore, the above artificial reverberation (artificial reverberation :) method is stimulated by a solid space to produce Reverb. Therefore, we established a reverb technology based on imaginary acoustic room. Generally speaking, this technique is based on a digital wave-ladder filter to simulate the propagation of one-dimensional sound waves. A virtual space is based on the geometric distribution of this space, which consists of several digital wave ladder filters, and simulates the situation of re-distribution of various vectors at the intersection of acoustic waves at various positions. The digital wave ladder filter and the spatial distribution matrix (Spatial Distribution) ^ 311 ^^ (801 ^)) form an artificial reverb generator, and the stability of this reverb generator can also be easily achieved, as long as these Filters and spatial distribution matrices are lossy from an energy point of view. This system is also very easy to adjust the system parameters without affecting the stability, such as the size of the virtual space and the gain of the reverberation. To produce real early reflections, we used finite impulse response (FIR) filters to simulate the signals before they entered the system filter network. 'Generally speaking, in the concept of this invention, it is just a collection of filter networks. The basic components of the filter network are derived from the one-dimensional waveform propagation model, and the spatial distribution matrix (SDM ) Is used to connect various basic components. The structure of this network is used to simulate the propagation of sound waves in a virtual acoustic space. The sound signal is input into a finite impulse response filter (FIR) to obtain early reflection. As a result, after inputting this filter network to obtain a long reverberation, the effect of reverb is generated, and this filter The stability of the network can also be proved. This paper scale is applicable to the Chinese National Standard (CNS) Α4ϋ ^ (210X297mm) (please read the notes on the back before filling in this page) 2. Description of the invention (屮) Generally speaking, each element of this invention is known to those in the field of digital signal processing and spatial acoustics. The imaginary room, ladder filter and spatial distribution matrices are also known to those who specialize in this field. Therefore, this invention combines these known techniques and techniques to obtain good reverberation (such as the sound of sweetness, stability, elasticity, etc.). Looking at the present invention from another angle, as long as the principle of wave propagation is used, a reverb generator that sounds pleasant can be easily generated. The invention can produce an artificial reverb that sounds more natural and substantial than the traditional way. It can be applied to the processing of sound signals, and can also be applied to the setting of home and commercial environments, such as theaters, concert halls, karaoke, etc. Additional benefits and features will become more apparent due to the more specific description and scope of patent applications proposed below. Brief description of the legend Figure 1 is a block diagram of a discrete-time system that generates a one-dimensional wave propagation. Printed and printed by the Beigong Consumer Cooperative of the Central Standards Bureau of the Ministry of Economic Affairs (please read the notes on the back to fill in this page) Figure 2 is a block diagram of a signal reflection and transmission at a junction. Figure 3 is a block diagram of one-dimensional wave conduction with multiple reflection intersections in a discrete-time mode. Figure 4 is a block diagram of a one-dimensional wave conduction practiced in discrete-time mode with multiple reflection junctions, plus the use of an attenuator to simulate a lossy signal path. This paper uses the Chinese National Standards (CNS) Α4 specification (21 Οχ 297 public) A7 B7 _ 5. Description of the invention (5) Figure 5 is a τ-dimensional wave with multiple reflection junctions practiced in discrete time Conduct and use a low-pass filter in the signal path to simulate a block diagram of an attenuated signal path. Figure 6 is a block diagram of a finite impulse response (FIR) filter to simulate 'early reflection. Figure 7 is a block diagram of an all_pass filter practiced with an infinite impulse response (IIR). Figure 8 is a block diagram of a comb filter practiced with an infinite impulse response (IIR). Figure 9 is a block diagram of a filter network reverb generator using a finite impulse response (FIR) filter 'followed by four wave ladder filters ° Figure 10 is a digital filter network reverb generator and it The functions supported by the side, and this is the practice of Figure ll.b. Figure Π.a and Figure 1 l.b are examples of virtual spaces that can be created by the filter network reverb generator. '-------- {install-- (please read the precautions on the back and then fill in this page), vs i; the detailed description of the invention printed by the Consumer Cooperative of the Central Standard Falcon Bureau of the Ministry of Economic Affairs. The embodiment is used as a reverb generator for post-processing of audio recording. The input sound signal is divided into two stages for processing: 'Finite Impulse Response (FIR) filter to simulate early reflection' and then a filter network to simulate long reverberation. The reverb generator is 'that is' based on the simulation of sound wave propagation. It converts continuous time (c ο ntin U 〇us -1 ime) -dimensional wave conduction into discrete time that can be practiced with the digital filter 10 of Fig. 1. model. Simulating wave conduction in digital form is known to those in the field of digital signal processing.-This paper standard is applicable to the Chinese National Standard (CNS) A4 specification (210X297 public jitter A7 B7 29S22d V. Description of the invention (6) And there are many books on this aspect that can be used as a reference 'such as LR Rabiner et al., Digital Processing of Speech Signals, Prentice-Hall, Inc., 1978. Figure 1 shows a simulated one-dimensional wave conduction without loss (Lossless) digital wave ladder filter 'it simulates the direct reflection of an object' like a wall '. The input is the original audio, and the output signal is the sound that returns to the original signal position after being reflected. This simple digital wave ladder filter includes Simulate the sound waves of the upper branch (upper branch) in the outward direction, and the lower branch (lower branch) in the direction of the sound wave reflected from the wall. Each branch is established by several delay modules 16 (delay module) The number of modules in each branch and the amount of delay in each module are determined by the size of the virtual acoustic space we want to simulate. A gain module 18 (ga in module) After connecting from the upper branch 12 to the lower branch M, it is equivalent to the amount of sound reflected back from the wall. Usually the amount of reflection will be less than 1, so the gain of the gain module will also be 1. The gain module The delay in the gain and delay module ^ in the program are programmable, so the user can adjust various parameters to match the characteristics of the sound. Since the sound wave can be moved from the end to the end_fine movement, This situation can be simulated in Figure 2 and Figure 3. Figure 2 shows that the lossless digital wave ladder has one segment-it is modeled by the empty_distribution matrix 24 (s as a model-dimensional wave_shenma Na, although there is no __ in the picture, the far _ wall _ seems to be represented by the upper part _ lower part of the right side of the reading path connected to the 1 benefit module. It is described in the previous figure. In this example, use Four sets of delay modules 20 (1), 2) are used, and the size of this paper is remotely used by China National Standard (CNS & A4 specifications (210X297mm) (please read the precautions on the back before filling out this page). The Central Falcon Bureau Beigong Consumer Cooperative printed the Ministry of Economic Affairs Central Standards Bureau employee consumption Cooperative cooperative printing A7 B7 5. Description of invention (7) 22 (1) and 22 (2). The delay module 20 (1) represents the transmission of sound waves from a pre-determined position to the space distribution matrix (SDM). The time spent between the represented contacts. The delay module 22 (1) represents the time it takes for the reflected sound waves to return from the spatial distribution matrix (SDM) through the same path as described above to the previously determined position. time. Therefore, the delays of modules 20 (1) and 22 (1) are indeed the same. It can be inferred that all delay modules' if they are located on the same acoustic path segment, they should have the same delay. It is used to simulate a sound wave passing through the junction of the same door, and then enter a space of different size, so part of the sound wave passes through the junction and part of the sound wave is reflected back. Similarly, the returned sound wave also undergoes the same process at the junction, that is, part of the sound wave returns to the main space through the junction, and part of the sound wave returns to the adjacent space. If the exact situation in the space can be accurately calculated, then the amount of reflection and transmission at each junction can be accurately determined. However, the actual situation in these spaces is quite difficult to simulate and measure. Therefore, a more practical approach is to generate a virtual acoustic space as described above, and then adjust the control parameters, such as the delay and coefficient of the space allocation matrix (SDM), to create the desired acoustic reverb effect. The Spatial Distribution Matrix (SDM) 24 in Figure 2 is simulated with diagonal coefficients (diag ο na 1 c 〇efficients) through the junction and coefficients (〇ff-diag ο na I c 〇efficients) that are not diagonal The case where the sound wave is reflected back from the junction, in this case it is a second-order (2X2) matrix. Figure 5 shows the case where three wave ladder filters as shown in Figure 4 are connected together. A paper scale applies to China National Standard (CNS) A4 specification (210X297mm) --------- <. Pack — (please read the notes on the back first and fill in this book)-Printed by the Staff Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs Install A7 B7. Fifth, the description of invention (8), this combination can be used to simulate multiple partially reflected objects in the acoustic path between the original contact and the wall, and each intermediate object reflects back part of the sound wave. The amount of delay is determined by the size of the virtual space to be simulated. In theory, the reflection coefficient can also be calculated from the characteristics of the virtual space, such as geometry and air density. However, in practice, it is very elastic, but because of this, how to produce good acoustic results is the main choice. Since acoustic energy usually attenuates gradually as it propagates in space, it is practical to add an attenuation factor to the wave ladder filter, as shown in Figures 4 and 5. Both Figures 4 and 5 show a one-dimensional wave conduction reflection represented by a spatial distribution matrix, and a propagation path between junctions plus an object with lossy (lossy). Except for adding a module for simulating energy attenuation in the acoustic path, Figure 3 is similar to Figures 4 and 5. In the example in Figure 4, the branch of the simulation outgoing sound wave is added with gain modules 30 (1), 30 (2) with gains +1, ^ 2, respectively, and the sound waves reflected back are simulated The branch also added gain modules 32 (1) and 32 (2) with gains §1 £ + 1 and §1 £ + 2, respectively. In the example of Figure 5, the filter modules 34 (1) and 34 (2) added to the branch, and the other modules added to the branch are the filter modules 36 (1) and 36 (2) . As shown in Figure 4, the gain modules at the same part of the acoustic path should have the same gain. Therefore, the same attenuation coefficient should be applied to the same part of the acoustic path, but the direction is opposite to the gain module -9 ~ This paper scale is applicable to the Chinese National Standard (CNS) (2 丨 0X2.97mm) (please Read the precautions on the back first and then fill out this page). Binding. Order ^ 93229 A7 B7 5. Description of the invention (closed)-··. In other words, the gain modules 30 (1) and 32 (1) should have the same value, and the gain modules 30 (2) and 32 (2) should also have the same value. In the lossy digital wave ladder filter shown in Figure 5, the filter can be a finite impulse response (FIR) filter as shown in Figures 7 and 8 or an infinite impulse response (IIR) filter . Both of these can be further elaborated and discussed in the current filter design book. Usually low-pass filter (low-pass filter) represents the absorption in the air. As we all know, high-frequency signals usually decay much faster than low-frequency signals. Of course, when designing a virtual space, users can not add The above parameters. However, analog low-frequency signals decay relatively quickly and it is understandable, as long as they sound good. In any design, the user can use any filter to control the attenuation rate. If the above wave ladder filter is applied to simulate early reflection (early reflection), it usually takes too much memory to build up the calculation amount. In order to reduce the calculation amount and memory, the finite impulse response (FIR) shown in Figure 6 ) The filter 40 can be used to simulate early reflections. The delay module 42 usually does not necessarily have the same delay amount and weight G for each delay, as shown in Figure 6, therefore, the output of the filter is a function of the input signal: output = Gmik m + Gm. iik.m + i + ...... + Gi ik.i + G〇ik The concept of early reflection is discussed in the book of Elements of Computer Mu sic by J. Moor. Usually ' Early reflection refers to those that are closer to the source body from the floor, wall or ____ _-10 -______ This paper scale uses the national standard Lee (CNS) A4 specification (210X297 mm) (please read the notes on the back first (Fill in this page again) Printed with the A7 B7 Employee Consumer Cooperative of the Ministry of Economic Affairs. Printed on the A7 B7 Employee Consumer Cooperative of the Central Sample Agency of the Ministry of Economic Affairs. 5. Description of the invention. () Sound waves reflected by other objects. Usually the delay of early reflection The time is distributed between 0 and 100 milliseconds. The determination of the proportion and delay in the finite impulse response filter (FIR filter) is known to those in the field of signal processing. So the details will not be described in detail However, usually the best way Directly measure the impulse response of a good acoustic space such as Carnegie Hall in New York City within 100 milliseconds, the measured impulse response can be used in this finite impulse response (FIR) filter, but this The method will be time-consuming and laborious. Figure 9 is a block diagram of a reverb built by the above digital wave ladder filter. The digitized input signal is processed by the finite impulse response (FIR) filter 50 to produce early reflections. (Early reflection) effect. This filtered signal is then input into the digital wave ladder filter 52 network to produce a long reverberation. In this embodiment, four digital wave ladder filters 52 are included Coupling via the spatial distribution matrix 54. The digital wave staircase filter is built according to the following principles. The spatial distribution matrix 54 combines the signals fed back from the wave staircase filter (for example, the signal labeled P 丨 j is U ..., 4), regenerate the output signal (for example, the signal labeled P 丨 'j is 1, ..., 4), and then combine it with the signal filtered by the finite impulse response (FIR) to generate separate digital waves The input signal of the step filter. The spatial distribution matrix (SDM) 54 calculates the sound wave pressure of the signal returned by each signal path simulated by the digital wave step filter. Then, the spatial distribution matrix (SDM) 54 calculates the output and then The sound wave pressure (eg P :) that is fed back to the digital wave ladder filter. -11--------- {袭 ------ book (please read the precautions on the back and fill in this page) This paper Wave scale General Chinese National Standard (CNS) A4 wash grid (2 丨 0> < 297mm) 293229 A7 B7 V. Description of invention (U) Space allocation matrix (SDM) 54 with four branches, which can be as follows Column matrix operation practice: ^

經濟部中央標準局負工消費合作社印製 其中P。,j = l,2,3,4 ’表示由空間分配矩陣的第j個路 徑傳播出去的信號,P丨,j = l,2,3,4,表示由空間分配矩 陣的第j個路徑輸入的信號。在此例中,J是一個4x4的矩 陣,空間分配矩陣的大小當然由輸入、輸出這矩陣的數目 所決定’空間分配矩陣的係數可以由一些已知聲學理論所 導出,或是可以隨便指定任意値。然而,爲了要達到穩定 性’這方法需要空間分配矩陣的特徵値(eigenvalues)的 量値(magnitude)以1爲上限。使用在數位波階梯濾波器 中的空間分配矩陣是爲模擬反射的交接點,亦有如上所示 的一般表示法。但典型的例子中有二組‘輸出、輸入的路 徑,因此,矩陣J是一個2x2的矩陣。 被衰減了 1/η(η=4,η爲數位波階梯濾波器的數目)的 有限脈衝響應(FIR)濾波器的輸出,再由加法器%加入每 個空間分配矩陣54的輸出,產生出各別數位波階梯濾波 器的輸入。另外一個加法器58,結合空間分配矩陣 (SDM)54的輸出聲壓形成了殘響的信號,另一衰減器 (attenuat〇r)60,帶有一可程式的增益値G,控制了殘響 信號的強度。 -· 12 - 本紙張尺度適用中國國家棣準(CNS ) A4規格(2丨0X297公釐) --------^ ·裝-- (請先閲讀背面之注意事項再填寫本頁)Printed by the Central Standards Bureau of the Ministry of Economic Affairs, the Consumer Labor Cooperative, where P. , J = l, 2, 3, 4 'represents the signal propagated by the jth path of the space allocation matrix, P 丨, j = l, 2, 3, 4 represents the input of the jth path of the space allocation matrix signal of. In this example, J is a 4x4 matrix, and the size of the space allocation matrix is of course determined by the number of input and output matrices. The coefficients of the space allocation matrix can be derived from some known acoustic theories, or can be arbitrarily specified. value. However, in order to achieve stability, this method requires the magnitude of the eigenvalues of the space allocation matrix to be 1 as the upper limit. The spatial distribution matrix used in the digital wave ladder filter is to simulate the intersection of reflections, and there is also a general notation as shown above. But a typical example has two sets of output and input paths. Therefore, the matrix J is a 2x2 matrix. The output of the finite impulse response (FIR) filter that has been attenuated by 1 / η (η = 4, η is the number of digital wave ladder filters) is added to the output of each spatial distribution matrix 54 by the adder%, resulting in The input of each digital wave ladder filter. Another adder 58, combined with the output sound pressure of the spatial distribution matrix (SDM) 54, forms a reverb signal, and another attenuator 60, with a programmable gain value G, controls the reverb signal Strength of. -· 12-This paper scale is applicable to China National Standard (CNS) A4 (2 丨 0X297mm) -------- ^ · Installation-- (Please read the precautions on the back before filling this page)

-、1T A7 A7 經濟部中央標準局員工消費合作社印製 五、發明説明(\7-) 數k波階梯濾波器的設計與建立方法並不是這麼蓽 要’然而它們被用來模擬一虛設聲學空間才是重點,而這 特別數位波階梯濾波器的設計就留給系統使用者去愼重地 決定了。當然’是由一些客觀的因素所決定,如系統聲音 的品質或特性等等。通常’要求聲音品質佳,這系統當然 就更加複雜了。除此之外,更複雜的系統給予使用者較大 的空間去g周整參數,使殘響產生器產生更適合的聲音。 圖十是一完整殘響產生器的具體範例,這系統是設計 去模擬如圖11. b所示具聲學特質的虛擬空間。圖十之數 位波階梯濾波器的數目相當於圖11.b用來模擬聲學空間的 分支的數目’本例子共有12個分支。聲音訊號62先由一 類比/數位轉換器64取樣之後變成有限脈衝響應(FIR)濾波 器的輸入信號。此電路的中心部份,除了在此例子中有 U個數位波階梯濾波器而不是像圖九中的四個之外,和 圖九所示是一樣的。實踐空間分配矩陣(SDM)的矩陣J, 在此例中,是一 1 2 X 1 2的矩陣。 最後,在衰減器60輸出之衰減過信號與代表直接聲響 訊號之原始信號經由一加法器68結合後,便輸入到一數 位/類比轉換器7 0產生最後輸出的類比信號。 參數控制單元72可以用..一微處理器或是已完成的硬體 去實踐’它控制數位波階梯濾波器的長度(也就是延遲 量),聲波的衰減量(例如增益),空間分配的樣本和殘響 ___ - 13 - 本紙张尺度適用中國國家標隼(CNS ) A4現格(210X297公釐) { -¾衣— (請先閲讀背面之注意事項再填寫本頁)-, 1T A7 A7 Printed by the Employee Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economy V. Description of invention (\ 7-) The design and establishment of the number-k wave staircase filter is not so important. However, they are used to simulate a dummy acoustic Space is the key point, and the design of this special digital wave ladder filter is left to the system user to make a heavy decision. Of course, it is determined by some objective factors, such as the quality or characteristics of the system sound. Normally, the sound quality is required, and the system is of course more complicated. In addition, a more complex system gives the user more room to round the parameters, making the reverb generator produce a more suitable sound. Figure 10 is a specific example of a complete reverb generator. This system is designed to simulate a virtual space with acoustic characteristics as shown in Figure 11.b. The number of Fig. 10 is the number of bit wave step filters equivalent to the number of branches used in Fig. 11.b to simulate acoustic space. In this example, there are 12 branches. The audio signal 62 is first sampled by an analog-to-digital converter 64 and becomes the input signal of a finite impulse response (FIR) filter. The central part of this circuit is the same as shown in Figure 9 except that there are U digital wave ladder filters in this example instead of the four in Figure 9. The matrix J of the practice space allocation matrix (SDM), in this example, is a 1 2 X 1 2 matrix. Finally, after the attenuated signal output from the attenuator 60 is combined with the original signal representing the direct sound signal through an adder 68, it is input to a digital / analog converter 70 to generate the final output analog signal. The parameter control unit 72 can be used: a microprocessor or a completed hardware to practice 'it controls the length of the digital wave ladder filter (that is, the amount of delay), the amount of attenuation of the sound wave (such as gain), space allocation Samples and remnants ___-13-This paper scale is applicable to the Chinese national standard falcon (CNS) A4 cash (210X297mm) {-¾ clothing — (Please read the precautions on the back before filling this page)

,1T 293229 A7 B7 五、發明説明(Θ) 經濟部中央標準局員工消費合作社印製, 1T 293229 A7 B7 V. Description of Invention (Θ) Printed by the Employee Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs

的衰減量。改變這些參數亦就是相當於改變虛擬空間的聲 學特性,但這間虛擬空間的基本形狀是沒有改變的。 設計的程序開始是建立虛擬空間的模型,如圖Ha.或 圖ll.b所示’在這些圖中的卽點就是根據上述的'—些原理 所設計而成的空間分配矩陣’而各個分支就代表的數位'波 階梯瀘波器。可將每一分支未端的交接點(node)代表原入 射方向訊號衰減後之純反射。 虛擬空間的模型可以根據一些敎科書所描述空間聲學 的原理而設計的’如John Wiley Co.出版的 .Fundamentals of Acoustics, Springer Verlag Co·出版 的 Concert Hall Acoustics和 Journal of Acoustics Society of America這本期刊中所發表的一些論文。虛擬 空間可以是任何形狀或大小的,而其聲學特性,如維度、 形狀或是否一活潑(lively)或殘響極少(dead)房間等’亦 是可選擇的。 ' 在每一個數位波階梯濾波器延遲單元的數目是由虛擬 空間分支的尺寸所決定的。增益因素(gain factor)通常說 定其強度小於1。若在波階梯濾波器中加入濾波器’爲^ 合能量在眞實空間環境中是逐漸衰減的事實低頻信號比闻 頻信號較不衰減,因此低通(l〇w_pass)或帶通(band-pass;) 濾波器是比較符合所需的。 另外記住幾個額外的規則對設計建立模型是很有幫 的’例如’若在系統中只有使用少數的波階梯濾波器’ G (請先閱讀背面之注意事項再填寫本頁) -装· '1ΤThe amount of attenuation. Changing these parameters is equivalent to changing the acoustic characteristics of the virtual space, but the basic shape of the virtual space remains unchanged. The design program starts with the establishment of a model of virtual space, as shown in Ha. Or ll.b. The last point in these figures is the space allocation matrix designed based on the above-mentioned principles. Represents the digital 'wave ladder wave filter. The end node of each branch can be used to represent the pure reflection of the original signal in the direction of attenuation after attenuation. The model of the virtual space can be designed according to the principles of spatial acoustics described in some books. As published by John Wiley Co. Fundamentals of Acoustics, Spring Hall Verlag Co., Concert Hall Acoustics and Journal of Acoustics Society of America. Some papers published in journals. The virtual space can be of any shape or size, and its acoustic characteristics, such as dimensions, shape or whether it is a lively or dead room, are also selectable. 'The number of delay elements in each digital wave ladder filter is determined by the size of the virtual space branch. Gain factor (gain factor) is usually said to be less than unity. If the filter is added to the wave ladder filter, the combined energy is gradually attenuated in the real space environment. The fact that the low frequency signal is less attenuated than the high frequency signal, so the low pass (l〇w_pass) or band pass (band-pass ;) The filter is more in line with the needs. Also remember a few additional rules are very helpful for designing the model. For example, "If only a few wave ladder filters are used in the system" G (please read the precautions on the back before filling this page)-install '1Τ

A7A7

的延遲量不應該太大,另外,在每一個分支內全部的延遲 量最好不要是其他分支的整數倍。若不遵從這規則,對於' 這相同延遲長度的某些頻率就會過份渲染、強調,便容易 產生不是想像中的結果。至於空間分配矩陣(SDM)內的 係數最好是小於1以確保系統的穩定性,爲了達到聽起來 是自然好聽的,最好是讓係數是隨意(random)的。但這 結果的影響不如前述分支延遲長度的影響來得大。 一個簡單的3度空間的聲學空間,是計算著4面牆壁、 地板及天花板所反射回來的信號,若每一面牆代表一個分 支,那至少需要六個分支,也就是說,空間分配矩陣 (SDM)的矩陣了,是一6x6的矩陣。 經濟部中央標準局員工消費合作杜印製 ---------1¾—— (請先閎讀背面之注意事項再填寫本頁) 訂 線 一個更複雜的虛擬空間模型舉例在圖11 中。在此例 中,在分支的交接點上,經由空間分配矩陣(SDM),鄰 近的分支就彼此互相連接。因此,空間分配矩陣 (SDM)80,將二個交接分支耦合著,產生傳送出去與反 射回來的信號,所以在空間分配矩陣(SDM)內之矩陣J爲 — 4x4的矩陣。然而,矩陣內的特徵値(eigenvalue)仍然 必須小於1。 必須澄淸的一點是,上述之技術是可以來模擬眞實的 聲學空間的。然而,與虛擬空間比較後發現在數位波階梯 濾波器的設計上或在變數的選擇上’眞實空間的模擬要更 加困難了。雖然前述之實施例爲數位化實踐方式,本發明 之電路亦是可用類比方式或混合方式來形成。當然’若採 -15 - 本紙張尺度適用中國國家標準(CNS ) A4規格(210X297公釐) A7 B7 五、發明説明(β) ~沪'· 用類比方式則就不需要圖十的類比/數位和數位/類比轉: 換器了。 其餘具體化的描述即包含在下列的專利範圍內。 裝 訂 線 (請先閲讀背面之注意事項再填寫本頁) 經濟部中央標隼局員工消f合作社印製 本紙張尺度適用中國國家標準(CNS ) A4規格(210X297公釐)The amount of delay should not be too large. In addition, the total amount of delay in each branch should not be an integer multiple of other branches. If this rule is not followed, certain frequencies with the same delay length will be over-rendered and emphasized, which will easily produce unimagined results. As for the coefficient in the space allocation matrix (SDM), it is better to be less than 1 to ensure the stability of the system. In order to achieve a natural and pleasant sound, it is best to let the coefficient be random. However, the effect of this result is not as great as the effect of the aforementioned branch delay length. A simple three-dimensional acoustic space is calculated from the signals reflected by the four walls, floor and ceiling. If each wall represents a branch, then at least six branches are needed, that is, the space allocation matrix (SDM) ) Is a 6x6 matrix. Du Printed by the Ministry of Economy, Central Bureau of Standards and Staff Consumption Cooperation --------- 1¾—— (please read the precautions on the back before filling out this page) An example of a more complex virtual space model is shown in Figure 11 in. In this example, at the junction of the branches, adjacent branches are connected to each other via a space allocation matrix (SDM). Therefore, the Space Allocation Matrix (SDM) 80 couples the two crossover branches to generate signals that are transmitted and reflected back, so the matrix J in the Space Allocation Matrix (SDM) is a 4x4 matrix. However, the eigenvalue in the matrix must still be less than 1. It must be clarified that the above techniques can be used to simulate the real acoustic space. However, comparing with virtual space, it is found that the simulation of real space is more difficult in the design of digital wave ladder filter or in the selection of variables. Although the foregoing embodiment is a digitized practical method, the circuit of the present invention can also be formed by an analog method or a hybrid method. Of course, if you adopt -15-this paper scale is applicable to the Chinese National Standard (CNS) A4 specification (210X297mm) A7 B7 V. Invention description (β) ~ Shanghai '· The analogy method does not require the analogy / digit of Figure 10. And digital / analog conversion: converter. The remaining specific descriptions are included in the following patent scope. Binding line (please read the precautions on the back before filling in this page) Printed by the Ministry of Economic Affairs, Central Standard Falcon Bureau Employee Consumer Cooperatives This paper size is applicable to the Chinese National Standard (CNS) A4 (210X297mm)

Claims (1)

經濟部中央標準局貝工消费合作社印袈 A8 B8 C8 D8 六、申_讎 1. 處理聲音訊_的:^波器網路殘響產生器,此裝置包含: 複數個波階梯濾波器(wave-ladder filter),其中每一 個皆是模擬在某一特定方向的波傳導相對傳送或反射的 聲音訊號,在那每一個上述複數個波階梯濾波器中,輸 入原始的聲音訊號,在各個方向產生不同的振盪,至少 有一個上述的波階梯濾波器必定包含一到數個次要 (secondary)空間分配矩陣(SDM)模組去模擬中間反射 的物體,而且一加法器結合上述所有的波階梯濾波器的 輸出信號以產生殘響的信號。 2. 如申請專利範圍第1項的濾波器網路殘響產生器,更含 一主要(primary)空間分配矩陣(SDM)模組,而此模組 將該複數個波階梯濾波器耦合起來,也就是說,接收上 述複數個波階梯濾波器的反射信爲輸入端·然後產生複 數個輸出信號,做爲波階梯濾波器的輸入= 3. 如申請專利範圍第1項所述的濾波器網路殘響產生器, 其中有一個以上的波階梯濾波器是由一至多個模擬中間 反射物體的次要(secondary)空間分配矩障(SDM)所組 合而成的。 4. 如申請專利範圍第2項所述的濾波器網路殘響產生器, 其中有--個以上的波階梯濾波器是由一至多個模擬中間 反射物體的次要(secondary)空間分配矩障(SDM)所組 合而成的。 本紙張尺度適用中國國家標準(CNS ) A4規格(210 X 297公釐) --------^1裝------訂-----《線 {請先閲讀背面之注**事項再填寫本頁) 經濟部中央標準局員工消費合作社印策 293229 as C8 D8 六、申請專利範圍 5. 如申請專利範圍第1項所述的濾波器網路殘響產生器, 該至少有一個的波階梯濾波器可以進一步加上一至數個 模擬在數位波階梯濾波器內反射聲音信號衰減量的增益 因素(gain factor)。 6. 如申請專利範圍第3項所述的濾波器網路殘響產生器, 該至少有一個的波階梯濾波器可以進一步加上一至數個 模擬在波階梯濾波器內反射聲音信號衰減量的增益因素 (gain factor) ° 7. 如申請專利範圍第4項所述的濾波器網路殘響產生器, 該至少有一個的波階梯濾波器可以進一步加上一至數個 模擬在波階梯濾波器內反射聲音信號衰減量的增益因素 (gain factor) ° 8. 如申請專利範圍第1、5、6、或7項所述的濾波器網路 殘響產生器,該至少有一個的波階梯濾波器可以進一步 加上一至數個模擬在波階梯濾波器內反射聲音與頻率大 小有關的濾波器。 9. 如申請專利範圍第8項所述的濾波器網路殘響產生器, 其中的爐波器可以由一群低通(low-pass)及帶通(bandpass) 濾波器中所選出。 10. 如申請專利範圍第2項所述的濾波器網路殘響產生器, 其中之主要空間分配矩陣和一至數個之次要空間分配矩 本紙張尺度適用中國國家標準(CNS ) A4規格(210 X 297公釐) ^ —裝 訂 (線 (請先閲讀背面之注意事項再填寫本頁) 經濟部中央標準局員工消費合作社印製 A8 B8 C8 D8 V'申請專利範圍 陣’皆係由一組邊界値小於1的特徵値(eigenvalue)所 代表。 U·如申請專利範圍第2項所述的濾波器網路殘響產生器, 更包含一個模擬聲音早期反射的有限脈衝響應(FIR)濾 波器。 12. 如申請專利範圍第2項所述的濾波器網路殘響產生器, 其中之複數個波階梯波器和主要空間分配矩陣(SDM) 模組,更包含可變元素(variable elements)來控制網 路的轉移特性(transfer characteristics)。 13. 如申請專利範圍第12項所述的濾波器網路殘響產生 器,更包含一參數控制單元(parameter control unit) 來控制在複數個波階梯濾波器及主要空間分配矩陣內 的多個可變元素(variable elements): 14. 如申請專利範圍第13項所述的濾波器網路殘響產生 器,其中之多個可變元素(variable elements)包括可 變延遲量的延遲模組。 1 5.如申請專利範圍第1 3項所述的濾波器網路殘響產生 器,其中之多個可變元素(variable elements)包括可 變增益的增益模組。 -19 - I ^1!^. I裝 訂 I 『 各 (請先Μ讀背面之注意事項再填寫本K ) 本紙伕尺度適用中國國家梯準(CNS ) Α4说格(210Χ297公釐〉 ABCD 293229 ~、申請專利範圍 16. 如申請專利範圍第12項所述的濾波器網路殘響產生 器,其中之主要空間分配矩陣模組更包括了複數個可 變係數(variable coefficients),而這些可變係數是由 參數控制單元(parameter control unit)所控制。 17. 如申請專利範圍第13項所述的濾波器網路殘響產生 器,更包括一模擬處理聲音早期反射的有限脈衝響應 (FIR)濾波器。 18. 如申請專利範圍第17項所述的濾波器網路殘響產生 器,更包括一增益模組,此模組用一可控制的增益因 素(gain factor)來放大或衰減殘響信號。 19. 如申請專利範圍第18項所述的濾波器網路殘響產生 器,更包括一第二加法器(adder),結合調整過增益的 殘響信號與原始聲音訊號以產生輸出的聲音訊號。 20. 如申請專利範圍第1項所述的濾波器網路殘響產生器· 可用數位方式建構之。 2 1.如申請專利範圍第1項所述的濾波器網路殘響產生器· 可用類比方式建構之。 本紙張尺度適用中國國家標準(CNS ) A4現格(210X297公釐) ^ —裝 訂 ( 線 (請先閱讀背面之注意事項再填寫本頁) 經濟部中央標率局負工消費合作社印製A8 B8, C8, D8, Beigong Consumer Cooperative, Central Standards Bureau, Ministry of Economic Affairs. A. B8, C8, D8. 1. Processing of audio signals: ^: Reverb generator for wave network, this device includes: a plurality of wave ladder filters (wave -ladder filter), each of which is a sound signal that simulates the relative transmission or reflection of the wave conduction in a specific direction. In each of the above-mentioned multiple wave ladder filters, the original sound signal is input and generated in all directions For different oscillations, at least one of the above wave ladder filters must contain one to several secondary space distribution matrix (SDM) modules to simulate objects that are intermediately reflected, and an adder combines all the above wave ladder filters The output signal of the generator is used to produce a reverb signal. 2. For example, the filter network reverb generator of item 1 of the patent scope further includes a primary space distribution matrix (SDM) module, and this module couples the plural wave ladder filters, In other words, receive the reflected signal of the complex wave ladder filter as the input terminal and then generate a plurality of output signals as the input of the wave ladder filter = 3. The filter network as described in item 1 of the patent application scope Road reverb generator, where more than one wave ladder filter is composed of one or more secondary space distribution moment barriers (SDM) that simulate intermediate reflective objects. 4. The filter network reverb generator as described in item 2 of the patent scope, in which more than one wave ladder filter is composed of one or more secondary space distribution moments that simulate an intermediate reflective object Barrier (SDM). The size of this paper is applicable to the Chinese National Standard (CNS) A4 specification (210 X 297 mm) -------- ^ 1 pack ------ order ----- "Line {Please read the back of the first (Note ** Please fill in this page again.) 293229 as C8 D8, the employee consumer cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs. 6. Patent application scope 5. The filter network reverb generator as described in item 1 of the patent application scope, the At least one wave ladder filter may further add one to several gain factors that simulate the attenuation of the reflected sound signal in the digital wave ladder filter. 6. The filter network reverb generator as described in item 3 of the patent application scope, the at least one wave ladder filter can be further added with one to several analogues of the attenuation of the reflected sound signal in the wave ladder filter Gain factor (gain factor) ° 7. The filter network reverb generator as described in item 4 of the patent scope, the at least one wave ladder filter can be further added with one to several analog wave ladder filters Gain factor of internal reflection sound signal attenuation amount 8. The filter network reverb generator as described in items 1, 5, 6, or 7 of the patent application scope, the at least one wave step filter The filter can be further added with one to several filters that simulate the frequency of the sound reflected in the wave-step filter. 9. The filter network reverb generator as described in item 8 of the scope of patent application, in which the oven can be selected from a group of low-pass (low-pass) and bandpass (bandpass) filters. 10. The filter network reverb generator as described in item 2 of the patent application scope, in which the main space allocation matrix and one to several secondary space allocation moments. This paper standard is applicable to the Chinese National Standard (CNS) A4 specification ( 210 X 297 mm) ^ —Binding (line (please read the notes on the back before filling in this page) A8 B8 C8 D8 V printed by the Consumer Cooperative of the Central Standards Bureau of the Ministry of Economic Affairs “Patent Application Array” is composed of a group The characteristic value (eigenvalue) whose boundary value is less than 1. U · The filter network reverb generator as described in item 2 of the patent application scope, further includes a finite impulse response (FIR) filter that simulates early reflection of sound 12. The filter network reverb generator as described in item 2 of the patent scope, in which a plurality of wave ladders and the main spatial distribution matrix (SDM) module further contain variable elements To control the transfer characteristics of the network. 13. The filter network reverb generator as described in item 12 of the patent application scope further includes a parameter control unit (parameter control unit) to control multiple variable elements in complex wave ladder filters and the main spatial distribution matrix: 14. The filter network reverb generator as described in item 13 of the patent scope, Among them, a plurality of variable elements (variable elements) include a delay module with a variable delay amount. 1 5. The filter network reverb generator as described in item 13 of the patent application scope, of which a plurality of variable elements Variable elements include variable gain gain modules. -19-I ^ 1! ^. I Binding I 『Each (please read the precautions on the back and then fill in this K)) The paper scale is applicable to China National Standards (CNS) Α4 said grid (210Χ297mm> ABCD 293229 ~, patent application scope 16. The filter network reverb generator described in item 12 of the patent application scope, the main space distribution matrix module includes A plurality of variable coefficients, and these variable coefficients are controlled by a parameter control unit. 17. The filter network reverb generator as described in item 13 of the patent scope, It includes a finite impulse response (FIR) filter that simulates the early reflection of sound. 18. The filter network reverb generator as described in item 17 of the patent application also includes a gain module, which uses a Controllable gain factor (gain factor) to amplify or attenuate the reverb signal. 19. The filter network reverb generator as described in item 18 of the patent application scope further includes a second adder, which combines the reverb signal adjusted with the gain and the original sound signal to generate the output sound signal . 20. The filter network reverb generator as described in item 1 of the patent scope can be constructed digitally. 2 1. The filter network reverb generator as described in item 1 of the patent application scope can be constructed by analogy. This paper scale is applicable to the Chinese National Standard (CNS) A4 (210X297mm) ^-binding (line (please read the precautions on the back before filling in this page) Printed by the Consumer Labor Cooperative of the Central Standardization Bureau of the Ministry of Economic Affairs
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