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TWI420918B - Low-complexity audio matrix decoder

Publication number

TWI420918B

TWI420918B TW95138970A TW95138970A TWI420918B TW I420918 B TWI420918 B TW I420918B TW 95138970 A TW95138970 A TW 95138970A TW 95138970 A TW95138970 A TW 95138970A TW I420918 B TWI420918 B TW I420918B

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TW

Taiwan

Prior art keywords

signals

signal

directional

program

right balance

Prior art date

2005-12-02

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• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
  • H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other

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• Engineering & Computer Science (AREA)
• Algebra (AREA)
• General Physics & Mathematics (AREA)
• Mathematical Analysis (AREA)
• Mathematical Optimization (AREA)
• Mathematical Physics (AREA)
• Pure & Applied Mathematics (AREA)
• Theoretical Computer Science (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• Stereophonic System (AREA)

本發明係有關於音訊信號處理。尤其是，本發明係有關於低複雜度適應性音訊矩陣解碼器或解碼程序，其皆適用於已編碼或未編碼之輸入信號之解碼。雖然其可用作獨立解碼器或解碼程序，但該解碼器或解碼程序亦可有益地與「虛擬器」或「虛擬化」程序來結合使用，使該解碼器或解碼程序對虛擬器或虛擬化程序提供多通道輸入。本發明亦有關儲存於電腦可讀取媒體上的電腦程式，以使電腦依據本發明之觀點執行解碼程序或解碼與虛擬化程序。The present invention relates to audio signal processing. In particular, the present invention relates to low complexity adaptive audio matrix decoders or decoding programs, all of which are suitable for decoding of encoded or uncoded input signals. Although it can be used as an independent decoder or decoder, the decoder or decoder can be advantageously used in conjunction with a "virtualizer" or "virtualization" program to make the decoder or decoder to virtual or virtual. The program provides multi-channel input. The invention also relates to a computer program stored on a computer readable medium for causing a computer to perform a decoding process or a decoding and virtualization program in accordance with the teachings of the present invention.

在此處所引用之每一個專利、公告專利申請案與參考文獻，其全文係於此被納入參考。Each of the patents, publications, and references cited herein are hereby incorporated by reference.

「虛擬耳機」與「虛擬喇叭」音訊處理器(「虛擬器」)典型上係將各與一方向相關聯之多聲道音訊信號編碼成兩個編碼聲道，使得當編碼聲道被施用至諸如一對耳機或一對喇叭之一對換能器時，針對該等換能器適當地被定位的聆聽者感覺該等音訊信號好像是來自與換能器不同之位置，而這些方向最好是與多聲道音訊信號的方向相關聯。耳機虛擬器典型上係形成使聆聽者感覺聲音是在「頭部外」而非頭部內的結果。虛擬耳機與虛擬喇叭處理器二者皆涉 及在其上施用之多聲道音訊信號的頭部相關轉換函數(HRTF)之運用。在此技術領域中，虛擬耳機與虛擬喇叭處理器係廣為人知的，並且其彼此類似(虛擬喇叭處理器可例如包括一「串音消除器」以與虛擬耳機處理器相異)。"Virtual Headset" and "Virtual Speaker" audio processors ("virtualizers") typically encode each multi-channel audio signal associated with a direction into two encoded channels such that when the encoded channel is applied to When a transducer such as a pair of headphones or a pair of speakers is paired, the listener appropriately positioned for the transducers senses that the audio signals appear to be from a different location than the transducer, and these directions are best Is associated with the direction of the multi-channel audio signal. The headset virtualizer is typically formed as a result of the listener feeling that the sound is "out of the head" rather than inside the head. Virtual headset and virtual speaker processor are involved And the use of a head related transfer function (HRTF) of the multi-channel audio signal applied thereto. In this technical field, virtual earphones and virtual speaker processors are well known and similar to each other (virtual speaker processors may include, for example, a "crosstalk canceller" to be different from a virtual earphone processor).

耳機與喇叭虛擬器之例包括以“Dolby Headphone”與“Dolby Virtual Speaker”為商標被販售之虛擬器。“Dolby”、“Dolby Headphone”與“Dolby Virtual Speaker”為Dolby Laboratories Licensing Corporation的商標。有關Dolby Headphone與Dolby Virtual Speaker之專利與申請案包括美國專利第6,370,256、6,574,649與6,741,706號及已公告之國際申請案WO 99/14983號。其他「虛擬器」包括例如在美國專利6449368號中所描述的以及在國際專利申請案WO 2003/053099號中所公告的。Examples of headphone and speaker virtualizers include virtual machines sold under the trademarks "Dolby Headphone" and "Dolby Virtual Speaker". "Dolby", "Dolby Headphone" and "Dolby Virtual Speaker" are trademarks of Dolby Laboratories Licensing Corporation. The patents and applications of the Dolby Headphone and the Dolby Virtual Speaker include U.S. Patent Nos. 6,370,256, 6,574,649 and 6,741,706, and the entire disclosure of the entire disclosure of the entire disclosure of the entire disclosures of Other "virtualizers" include those disclosed in, for example, U.S. Patent No. 6,449,368, the disclosure of which is incorporated herein by reference.

Dolby Headphone與Dolby Virtual Speaker分別提供使用一對標準耳機或一對標準喇叭之多聲道環繞音響的感覺。最近低複雜度版本之Dolby Headphone與Dolby Virtual Speaker已問世，其例如在廣泛的各種新的低成本產品中為有用處的，諸如多媒體行動電話、可攜式媒體播放器、可攜式遊戲機與低成本電視機。然而此類低成本產品典型上為二聲道立體音效(立體聲)裝置，而虛擬器需要多聲道環繞音響輸入。The Dolby Headphone and Dolby Virtual Speaker offer the feeling of using a pair of standard headphones or a pair of standard speakers for multi-channel surround sound. Recently, low-complexity versions of the Dolby Headphone and Dolby Virtual Speaker have been introduced, for example, in a wide variety of new low-cost products such as multimedia mobile phones, portable media players, portable game consoles and Low cost TV. However, such low cost products are typically two-channel stereo sound (stereo) devices, while virtual devices require multi-channel surround sound input.

雖然例如Dolby Pro Logic II與其前身Pro Logic之現有矩陣解碼器在將低成本裝置之二聲道立體聲音訊輸出媒配為Dolby Headphone虛擬器之多聲道環繞音響輸入上為有 用處的，但在一些低成本裝置的使用上，現有的矩陣解碼器一般來說可能會比大家希望的還要更複雜且資源密集性更高。“Dolby Pro Logic”與“Dolby Pro Logic II”為Dolby Laboratories Licensing Corporation之商標。Dolby Pro Logic II之觀點係於美國專利第6,920,223號與6,970,567號及國際專利申請案WO 2002/019768號中提出。Dolby Pro Logic之觀點係於美國專利第4,799,260號、4,941,177與5,046,098號中提出。Although, for example, Dolby Pro Logic II and its predecessor, Pro Logic's existing matrix decoder, have a multi-channel surround sound input that combines the two-channel stereo audio output of a low-cost device with the Dolby Headphone virtual device. Useful, but in the use of some low-cost devices, existing matrix decoders may generally be more complex and resource-intensive than desired. "Dolby Pro Logic" and "Dolby Pro Logic II" are trademarks of Dolby Laboratories Licensing Corporation. The idea of Dolby Pro Logic II is set forth in U.S. Patent Nos. 6,920,223 and 6,970,567, the entire disclosure of which is incorporated herein by reference. The views of Dolby Pro Logic are set forth in U.S. Patent Nos. 4,799,260, 4,941,177 and 5,046,098.

因而，對低複雜度矩陣解碼器之需求係存在的，尤其是要用在虛擬器上及為了在虛擬器上使用而改良的矩陣解碼器，特別地係對於諸如Dolby Headphone與Dolby Virtual Speaker之虛擬器者。理想上，此種矩陣解碼器應使每一程序階段的複雜度最小化，以獲得類似Dolby Pro Logic II解碼器之績效。Thus, the need for low complexity matrix decoders exists, especially for matrix decoders that are used on virtual machines and for use on virtual machines, especially for virtual applications such as Dolby Headphone and Dolby Virtual Speaker. The device. Ideally, such a matrix decoder should minimize the complexity of each program stage to achieve performance similar to the Dolby Pro Logic II decoder.

本發明係有關於一種用於處理音訊信號之方法，其包含：(1)由m個音訊輸入信號導出n個音訊輸出信號，此處m與n為正整數，且該等n個音訊輸出信號係使用響應於一個或多個控制信號之一適應性矩陣或矩陣化程序來導出，此矩陣或矩陣化程序在響應m個音訊信號下產生n個音訊信號；以及(2)由該等m個音訊輸入信號導出多個隨時間變化之控制信號，其中該等控制信號係使用以下裝置或程序而由該等m個音訊輸入信號來導出：(a)在響應於該等m個音訊 輸入信號下產生多個方向性支配信號之一處理器或程序，其中至少一個方向性支配信號係與一第一方向軸有關，且至少另一個方向性支配信號係與一第二方向軸有關；以及(b)在響應於該等方向性支配信號下產生該等控制信號之一處理器或程序。The invention relates to a method for processing an audio signal, comprising: (1) deriving n audio output signals from m audio input signals, where m and n are positive integers, and the n audio output signals Deriving using an adaptive matrix or matrixing procedure responsive to one or more control signals, the matrix or matrixing process generating n audio signals in response to m audio signals; and (2) by the m The audio input signal derives a plurality of time varying control signals, wherein the control signals are derived from the m audio input signals using: (a) in response to the m audio signals a processor or program for generating a plurality of directional dominant signals under the input signal, wherein at least one directional dominant signal is associated with a first directional axis and at least one other directional dominant signal is associated with a second directional axis; And (b) generating a processor or program of the control signals in response to the directional dominant signals.

該適應性矩陣或矩陣化程序可包括：(1)一被動矩陣或矩陣化程序，其在響應於該等m個音訊信號下產生n個音訊信號；(2)振幅比例調整器或振幅比例調整程序，其各在響應於一時變振幅比例調整因子控制信號下將該被動矩陣或矩陣化程序所產生的其中一個控制信號作振幅比例調整，以產生該等n個音訊輸出信號，其中該等多個時變控制信號為n個時變振幅比例調整因子控制信號，為用於將該被動矩陣或矩陣化程序所產生的每一個音訊信號作振幅比例調整者。The adaptive matrix or matrixing program may include: (1) a passive matrix or matrixing process that generates n audio signals in response to the m audio signals; (2) amplitude scale adjuster or amplitude ratio adjustment And each of the control signals generated by the passive matrix or the matrixing process is amplitude-scale adjusted in response to the one-time variable amplitude scaling factor control signal to generate the n audio output signals, wherein the plurality of The time varying control signals are n time varying amplitude scaling adjustment factor control signals, which are used to adjust the amplitude ratio of each audio signal generated by the passive matrix or matrixing program.

m值可為2，且n值可為4或5。The m value can be 2, and the value of n can be 4 or 5.

產生方向性支配信號之處理器或程序可使用：(1)在響應於該等m個音訊輸入信號下產生多對信號之一被動處理器或程序，其第一對信號代表沿著一第一方向軸之相對向方向的信號強度，且第二對信號代表沿著一第二方向軸之相對向方向的信號強度；以及(2)在響應於該等二對信號下產生該等多個方向性支配信號之一處理器或程序，至少一個方向性支配信號係與該等第一及第二方向軸中之一個相關。A processor or program for generating a directional dominant signal may use: (1) a passive processor or program that generates a plurality of pairs of signals in response to the m audio input signals, the first pair of signals representing a first a signal strength of the opposite direction of the direction axis, and the second pair of signals represents signal strengths in opposite directions along a second direction axis; and (2) generating the plurality of directions in response to the two pairs of signals A processor or program of one of the dominant dominant signals associated with one of the first and second directional axes.

產生多個方向性支配信號之該處理器或程序可使用獲 得每一對信號的量值間之正或負差的線性振幅域減除器或減除程序、放大每一個該等差之放大器或放大程序、實質地將每一個被放大之差限制於一正截波位準與一負截波位準上之截波器或截波程序、以及將每一個被放大與限制之差作時間平均動作之平滑器或平滑程序。The processor or program that generates multiple directional dominant signals can be used A linear amplitude domain subtractor or subtraction procedure that produces a positive or negative difference between the magnitudes of each pair of signals, amplifies each of the equalization amplifiers or amplification procedures, and substantially limits each amplified difference to one A chopper or a chopping program at the positive chopping level and a negative chopping level, and a smoother or smoothing procedure for time-averaging the difference between each of the amplified and limited.

產生多個方向性支配信號之該處理器或程序可使用獲得每一對信號的量值間之正或負差的線性振幅域減除器或減除程序、實質地將每一個被放大之差限制於一正截波位準與一負截波位準上之截波器或截波程序、放大每一個該等差之放大器或放大程序、以及將每一個被放大與限制之差作時間平均動作之平滑器或平滑程序。The processor or program that generates the plurality of directional dominant signals may use a linear amplitude domain subtractor or subtraction procedure that obtains a positive or negative difference between the magnitudes of each pair of signals, substantially the difference between each amplified Limiting the chopping or clipping process at a positive chopping level to a negative chopping level, amplifying each of the equalizing amplifiers or amplification procedures, and time-averaging the difference between each amplified and limited Motion smoother or smoothing program.

放大器或放大程序之放大因子與截波器或截波程序功能用來限制放大差的截波位準間的關係可構成量值之正與負臨界值，低於此臨界值的被限制及放大之差可能會具有介於0與實質上該截波位準間的振幅，且高於此臨界值的被限制及放大之差可具有實質上為此截波位準之振幅。The amplification factor of the amplifier or amplification program and the relationship between the chopping or clipping program function used to limit the clipping level of the amplification difference may constitute positive and negative threshold values of the magnitude, below which the limit value is limited and amplified. The difference may have an amplitude between 0 and substantially the chopping level, and the difference between the limited and amplified values above the threshold may have an amplitude that is substantially the chopping level for this.

就沒有相關之音訊輸入信號而言，方向性支配信號可能會基於比較多對信號之比值而近似一方向性支配信號，而就相關之音訊輸入信號而言，方向性支配信號可能會傾向負或正的截波位準。In the absence of an associated audio input signal, the directional dominant signal may approximate a directional dominant signal based on the ratio of the more pairs of signals, and the directional dominant signal may tend to be negative or related to the associated audio input signal. Positive chopping level.

被限制及放大之差針對該差的變換函數來說，在該等臨界值間實質上可為線性的。The difference between the limit and the amplification is substantially linear between the threshold values for the difference transfer function.

高於正臨界值之差可指出沿著一方向軸的正支配，而低於負臨界值之差可指出沿著一方向軸的負支配，且介於 正與負臨界值間之差可指出沿著一方向軸的非支配。A difference above a positive threshold may indicate a positive dominance along a direction axis, and a difference below a negative threshold may indicate a negative dominance along a direction axis, and The difference between positive and negative thresholds can indicate non-domination along one direction axis.

產生多個方向性支配信號之處理器或程序亦可在平滑動作之前或之後修改該被放大及限制之差，使得導出之方向性支配信號在該與方向性支配信號相關的軸上被偏置。The processor or program that generates the plurality of directional dominant signals may also modify the amplified and limited difference before or after the smoothing action such that the derived directional dominant signal is biased on the axis associated with the directional dominant signal .

產生多個方向性支配信號之處理器或程序亦可在沿著方向軸有非支配與有正或負支配時差別地修改該被放大及限制之差。The processor or program that produces the plurality of directional dominant signals can also modify the difference between the amplification and the limit differentially when there is non-domination and positive or negative dominance along the direction axis.

在響應於該等多個方向性支配信號下產生該等控制信號之處理器或程序可施用至少一個左右平衡函數(panning function)至每一個該等多個方向性支配信號。A processor or program that generates the control signals in response to the plurality of directional dominant signals may apply at least one left and right balance function to each of the plurality of directional dominant signals.

在另一觀點中，本發明可由該等n個音訊輸出信號導出p個音訊信號，其中p為2且該等p個音訊信號係使用虛擬器或虛擬程序而自該等n個音訊信號導出，使得當該等p個音訊信號被施用至一對換能器時，針對該等換能器適當地被定位之一聆聽者感覺該等n個音訊信號好像來自與該等換能器不同的位置。該虛擬器或虛擬程序可包括施用一個或多個頭部相關轉換函數至該等n個音訊輸出信號中之數個信號。該等換能器可為一對耳機或一對喇叭。In another aspect, the present invention can derive p audio signals from the n audio output signals, where p is 2 and the p audio signals are derived from the n audio signals using a virtualizer or virtual program. Such that when the p audio signals are applied to a pair of transducers, one of the listeners is appropriately positioned for the transducers to feel that the n audio signals appear to be from a different location than the transducers . The virtualizer or virtual program can include applying one or more head related conversion functions to the plurality of the n audio output signals. The transducers can be a pair of headphones or a pair of speakers.

雖然本發明之觀點在其他型式的矩陣解碼器來上亦可適用，但在一例示性實施例中，採用了固定矩陣可變增益做法，原因為其與可變矩陣做法相較之下的較低複雜度。使用可變增益解碼器所產生的單一音源之優異的隔離性在常為單音情況的遊戲聲中是可接受的，如果沒有更好的選擇。While the present invention is also applicable to other types of matrix decoders, in an exemplary embodiment, a fixed matrix variable gain approach is employed for reasons that are comparable to variable matrix practices. Low complexity. The superior isolation of a single source produced using a variable gain decoder is acceptable in game sounds that are often monophonic, if there is no better choice.

在配合虛擬器工作時，會想要盡可能地降低聲道間洩漏，原因在於不同聲道之頭部相關轉換函數(HRTF)間的相互作用與扺銷。可變增益做法允許完全關閉某些聲道而維持最小的聲道間洩漏。When working with a virtualizer, you want to minimize inter-channel leakage because of the interaction and credit between the head-related transfer functions (HRTF) of different channels. The variable gain approach allows some channels to be completely turned off while maintaining minimal inter-channel leakage.

進一步言之，在使用可變增益解碼器時，於某些信號狀況中會發生之「泵動」副作用不會像配合虛擬器來使用時那麼令人討厭。此乃由於虛擬器的每一個輸入聲道產生二聲道之輸出的性質所致。雖然可變增益之矩陣解碼器會造成某些喇叭完全關閉，但只要其至少有一個輸入為開啟的，那麼一虛擬器之二輸出就都不會被關閉。Furthermore, when using a variable gain decoder, the "pumping" side effects that occur in certain signal conditions are not as annoying as when used with a virtual machine. This is due to the nature of the output of the two channels produced by each input channel of the virtual machine. Although the variable gain matrix decoder will cause some speakers to be completely turned off, as long as at least one of the inputs is turned on, the output of one of the virtual devices will not be turned off.

如下面進一步解釋地，最佳化可能是為了程序可變增益做法之另一個已知的缺點-漏失非支配信號而做的，結果形成兼具有這二個世界之最佳功能的解碼器。As explained further below, optimization may be done for another known disadvantage of the program variable gain approach - missing non-dominated signals, resulting in a decoder that combines the best of both worlds.

同時，由於依據本發明之觀點的矩陣解碼器之一用途為替虛擬器導出多聲道內容，故輸出之個數可被限制為4個：左、右、左環繞與右環繞。事實上，虛擬器之主要目標為要輸送所有圍繞著聆聽者的良好感覺方向性，此可只使用四聲道而省略中央聲道來達成，中央聲道之納入會顯著地增加處理執行時間，而只邊緣性地加強方向性之感知。Meanwhile, since one of the matrix decoders according to the viewpoint of the present invention is to derive multi-channel content for the virtual device, the number of outputs can be limited to four: left, right, left surround, and right surround. In fact, the main goal of the virtual machine is to convey all the good directionality around the listener, which can be achieved by omitting the center channel using only four channels. The inclusion of the center channel can significantly increase the processing execution time. It only marginally enhances the perception of directionality.

由於頭部相關轉換函數(HRTF)被相加在一起時會產生破壞性干擾，因此較佳的是避免聲道間的相關。換言之，虛擬器在當音源儘可能多地一次朝向一個喇叭操縱時會表現地較佳。然而，此一結果之達成應在整體聲音階級之折衷上取得平衡。Since the head related conversion functions (HRTFs) are added together to cause destructive interference, it is preferable to avoid correlation between channels. In other words, the virtualizer will perform better when the sound source is manipulated as much as possible toward one horn. However, the achievement of this result should be balanced in the compromise of the overall sound class.

第1圖為示意性功能方塊圖，顯示依據本發明之觀點，用於由多個音訊輸入信號導出多對中間控制信號的處理器或程序之一例，其中該等多對中間控制信號代表沿著一方向軸之相對向方向的信號強度。在此例中，其可被指定為「第一階段」，其有二個音訊輸入信號Lin與Rin，且其有二對中間控制信號L-R與F-B。1 is a schematic functional block diagram showing an example of a processor or program for deriving a plurality of pairs of intermediate control signals from a plurality of audio input signals in accordance with the teachings of the present invention, wherein the plurality of pairs of intermediate control signals are representative along The signal strength of the opposite direction of a direction axis. In this example, it can be designated as the "first stage", which has two audio input signals Lin and Rin, and has two pairs of intermediate control signals L-R and F-B.

第2圖為示意性功能方塊圖，顯示依據本發明之觀點，用於導出多個方向性支配信號之處理器或程序的一例，針對每一對中間控制信號至少導出一個此種信號。在此例中，其可被指定為「第二階段」，其有二對中間控制信號L-R與F-B及二個方向性支配信號LR與LB。2 is a schematic functional block diagram showing an example of a processor or program for deriving a plurality of directional dominant signals in accordance with the teachings of the present invention, at least one such signal being derived for each pair of intermediate control signals. In this example, it can be designated as the "second stage", which has two pairs of intermediate control signals L-R and F-B and two directional dominant signals LR and LB.

第3圖顯示以正交的LR與FB軸為基礎之二維平面中的觀念性或理論性方向支配向量的一例。Figure 3 shows an example of a conceptual or theoretical directional dominating vector in a two-dimensional plane based on orthogonal LR and FB axes.

第4圖為信號振幅對上時間之理想化圖示，其分別顯示一個二聲道立體聲信號的絕對值L與R，其中左輸入聲道(Lin)在取絕對值前為具有0.4的尖峰振幅之50Hz正弦波，而右輸入聲道(Rin)在取絕對值前為具有(50*)Hz的頻率及1.0的尖峰振幅之正弦波。該等正弦波之頻率為不相關的，而左聲道之位準為右聲道之位準的0.4倍。Figure 4 is an idealized representation of the signal amplitude versus time, which shows the absolute values L and R of a two-channel stereo signal, respectively, where the left input channel (Lin) has a peak amplitude of 0.4 before taking the absolute value. 50Hz sine wave, and the right input channel (Rin) has (50*) before taking the absolute value A sine wave with a frequency of Hz and a peak amplitude of 1.0. The frequencies of the sinusoids are irrelevant, and the level of the left channel is 0.4 times the level of the right channel.

第5圖為信號振幅對上時間之理想化圖示，顯示由R減掉L，以及乘上差然後在-1.0與1.0截波以提供一準長方形波二者之結果。Figure 5 is an idealized representation of the signal amplitude versus time, showing the result of subtracting L from R and multiplying the difference and then chopping at -1.0 and 1.0 to provide a quasi-rectangular wave.

第6圖為信號振幅對上時間之理想化圖示，顯示由將該 準長方形波饋送穿過一平滑器濾波器所致之平滑後的LR中間控制信號，其說明了，對於實質上無相關之信號輸入，方向性支配信號會趨近於靠近沿著與LR中間控制信號相關的方向軸之信號強度的比值式比較結果之值。Figure 6 is an idealized representation of the signal amplitude versus time, shown by The quasi-rectangular wave feeds through the smoothed LR intermediate control signal caused by a smoother filter, which illustrates that for substantially uncorrelated signal input, the directional dominant signal will approach the control along the middle of the LR The value of the ratio comparison of the signal strengths of the signal-related direction axes.

第7圖為示意式功能方塊圖，顯示依據第2圖顯示之本發明的觀點之處理器或程序的修正。在此例中，此亦可被指定為「第二階段」，被放大及截波之BF差被限制為小於0的值以將FB支配信號向後偏置。Figure 7 is a schematic functional block diagram showing the modification of the processor or program in accordance with the teachings of the present invention shown in Figure 2. In this example, this can also be designated as the "second phase", and the BF difference of the amplified and chopped is limited to a value less than 0 to bias the FB dominating signal backward.

第8圖為以弧線表示的增益對上角度之理想化圖示，顯示左(L)與右(R)音頻聲道間之共同左右平衡法則(pan-law)，即正弦/餘弦左右平衡法則，此處L=cos(x)*input，且R=sin(x)*input，而x在0至π/2間變化。Figure 8 is an idealized representation of the gain versus upper angle in arcs, showing the common left-right balance rule (pan-law) between the left (L) and right (R) audio channels, ie the sine/cosine left and right balance rule Where L = cos(x) * input, and R = sin(x) * input, and x varies between 0 and π/2.

第9a圖為當第8圖中之相同的正弦/餘弦左右平衡法則被施用至LR軸時，分別就panL與panR顯示增益對上方向性支配信號位準之理想化圖示，panL與panR分別代表來自左與右之增益分佈。Figure 9a shows an idealized representation of the gain versus the directional dominant signal level for panL and panR when the same sine/cosine left and right balance rule is applied to the LR axis in Figure 8, respectively. Represents the gain distribution from left and right.

第9b圖為當第8圖中之相同的正弦/餘弦左右平衡法則被施用至FB軸時，分別就panB與panF顯示增益對上方向性支配信號位準之理想化圖示，panB與panF分別代表來自後與前之增益分佈。Figure 9b shows an idealized representation of the gain versus the directional dominant signal level for panB and panF when the same sine/cosine left and right balance rule is applied to the FB axis in Figure 8, respectively. Represents the gain distribution from the back and the front.

第10圖為一理想化圖示，顯示LGain公式之準三維呈現(其三軸為正規化之增益及FB與LR之值)。Figure 10 is an idealized diagram showing the quasi-three-dimensional representation of the LGain formula (its three axes are the normalized gain and the values of FB and LR).

第11圖為一理想化圖示，顯示LGain，RGain，LsGain與RsGain公式之準三維呈現(其三軸為正規化之增益及FB 與LR之值)。Figure 11 is an idealized diagram showing the quasi-three-dimensional representation of the LGain, RGain, LsGain and RsGain formulas (the three axes are the normalized gain and FB). With the value of LR).

第12圖為一理想化圖示，顯示一餘弦波與介於0及π/2間之餘弦的第二階多項式近似，其顯示在0<x<1之範圍內，y=(1-x² )之近似式合理地接近於y=cos(x* π/2)。圖中下面之曲線為該近似式。Figure 12 is an idealized diagram showing a second-order polynomial approximation of a cosine wave with a cosine between 0 and π/2, which is shown in the range of 0 < x < 1, y = (1-x ² ) The approximation is reasonably close to y=cos(x* π/2). The lower curve in the figure is the approximation.

第13圖為一理想化圖示，顯示LGain，RGain，LsGain與RsGain公式之準三維呈現(其三軸為正規化之增益及FB與LR之值)。其中在計算LGain與RGain時，LR左右平衡分量未被使用。Figure 13 is an idealized diagram showing the quasi-three-dimensional representation of the LGain, RGain, LsGain and RsGain formulas (the three axes are the normalized gain and the values of FB and LR). When calculating LGain and RGain, the left and right balance components of LR are not used.

第14圖為示意式功能方塊圖，顯示依據本發明的之觀點，用於由多個方向性支配信號導出多個控制信號的處理器或程序之一例。在可被指定為「第三階段」的此例中，四個控制信號LGain，RGain，LsGain與RsGain係由二個方向性支配信號LR與FB來導出。Figure 14 is a schematic functional block diagram showing an example of a processor or program for deriving a plurality of control signals from a plurality of directional dominant signals in accordance with the teachings of the present invention. In this example, which can be designated as the "third stage", the four control signals LGain, RGain, LsGain and RsGain are derived from the two directional dominant signals LR and FB.

第15圖為示意式功能方塊圖，顯示依據本發明之觀點，用於由音訊輸入信號與多個控制信號導出多個音訊輸出信號的適應性矩陣或矩陣化程序之一例。在可被指定為「第四階段」的此例中，一對音訊輸入信號Lin與Rin被施用至一被動矩陣，並且每一個矩陣輸出的位準係由四個控制信號LGain、RGain、LsGain與RsGain之一各別來控制的，以產生四個音訊輸出信號LOut、POut、LsOut與RsOut。Figure 15 is a schematic functional block diagram showing an example of an adaptive matrix or matrixing procedure for deriving a plurality of audio output signals from an audio input signal and a plurality of control signals in accordance with the teachings of the present invention. In this example, which can be designated as the "fourth stage", a pair of audio input signals Lin and Rin are applied to a passive matrix, and the level of each matrix output is controlled by four control signals LGain, RGain, LsGain and One of the RsGains is controlled separately to generate four audio output signals LOut, POut, LsOut and RsOut.

第16圖為一示意式方塊圖，顯示本例全部四個階段的綜述，並指出其相互關係。Figure 16 is a schematic block diagram showing an overview of all four phases of this example and indicating their relationship.

本發明之觀點可配合例示性實施例較佳地被了解，此實施例為了方便描述而被分為四個「階段」。此四個階段在接收m個音訊輸入信號、於此例中為Lin與Rin的兩個信號，以及輸出n個音訊信號、於此例中為LOut(左輸出)、ROut(右輸出)、LsOut(左環繞輸出)與RsOut(右環繞輸出)的四個信號，的適應性音訊解碼器或解碼程序之內文中的整體關係於第16圖中顯示。該解碼器或解碼程序具有一控制路徑，其包括第一、第二、第三階段，及包括在第四階段中之適應性矩陣或矩陣化程序的一信號路徑。在此例中，多個時間變化控制信號、該控制路徑產生四個控制信號且此等信號被施用至該適應性矩陣或矩陣化程序。The aspects of the present invention are best understood in conjunction with the exemplary embodiments, which are divided into four "stages" for ease of description. The four stages receive m audio input signals, in this case two signals of Lin and Rin, and output n audio signals, in this case LOut (left output), ROut (right output), LsOut The overall relationship in the context of the adaptive audio decoder or decoding program for the four signals (left surround output) and RsOut (right surround output) is shown in Figure 16. The decoder or decoder has a control path that includes first, second, third stages, and a signal path including an adaptive matrix or matrixing procedure in the fourth stage. In this example, a plurality of time varying control signals, the control path produces four control signals and the signals are applied to the adaptive matrix or matrixing procedure.

看向第1圖中所顯示之第一階段，在此例中，m個音訊輸入信號Lin與Rin被施用至一處理器或程序，其在響應於該等m個音訊輸出信號下導出多對信號，即，代表沿著一第一方向軸(在此例中為L-R或左-右軸)之相對向的信號強度的第一對信號L與R，與代表沿著一第二方向軸(在此例中為F-B或前-後軸)之相對向的信號強度的第二對信號F與B。雖然此例運用二正交方向軸，但也可有多於二個的方向軸(且因而在沿著各個額外方向軸上會有多於二對代表相對向的信號強度之信號)，且該等軸未必為正交的(如，見美國專利第6,970,567號)。第一階段之處理器或程序可被視為被動矩陣或矩陣化程序。在此例中，一簡單的矩陣計算左、右、 和與差信號，且使用其絕對值以作為中間控制信號L，R，F與B。更明確地說，在此例中之被動矩陣或被動矩陣化程序可用下列公式來特徵化：L=|Lin|Looking at the first stage shown in Figure 1, in this example, m audio input signals Lin and Rin are applied to a processor or program that derives multiple pairs in response to the m audio output signals a signal, i.e., a first pair of signals L and R representing the relative signal strength along a first direction axis (in this case LR or left-right axis), and a representative along a second direction axis ( In this example, the second pair of signals F and B of the relative signal strength of the FB or the front-rear axis). Although this example utilizes two orthogonal direction axes, there may be more than two direction axes (and thus there may be more than two pairs of signals representing the relative signal strength along the respective additional direction axes), and The equiaxions are not necessarily orthogonal (see, for example, U.S. Patent No. 6,970,567). The first stage processor or program can be considered a passive matrix or matrix program. In this case, a simple matrix calculates left and right, The sum and difference signals, and their absolute values are used as intermediate control signals L, R, F and B. More specifically, the passive matrix or passive matrixing procedure in this example can be characterized by the following formula: L=|Lin|

R=|Rin|R=|Rin|

F=|(0.5*Lin)+(0.5*Rin)|F=|(0.5*Lin)+(0.5*Rin)|

B=|(0.5*Lin)-(0.5*Rin)|B=|(0.5*Lin)-(0.5*Rin)|

現在看向第2圖中所顯示之第二階段，多對信號(每一對信號代表沿著一方向軸的相對向之信號強度)被施用至產生多個方向性支配信號之一處理器或程序。在此例中，有二對信號L-R與F-B被施用至第二階段，且有二個方向性支配信號由第二階段產生。原則上如上面所提及的，其可能會有多於二個的方向軸(且因而有多於二對的信號及多於二個的方向性支配信號)。也可能會產生比這些信號對以及相關軸更多的方向性支配信號。這有可能會由以多於一個方式處理一對被施用之信號以在響應於一特定對的被施用之信號下產生多重方向性支配信號來達成。在看向第二階段之範例的細節前，先解釋第二階段之操作原理是有用處的。Looking now at the second stage shown in Figure 2, multiple pairs of signals (each pair of signals representing the relative signal strength along a direction axis) are applied to one of the processors that generate multiple directional dominant signals or program. In this example, two pairs of signals L-R and F-B are applied to the second stage, and two directional dominant signals are generated by the second stage. In principle, as mentioned above, there may be more than two directional axes (and thus more than two pairs of signals and more than two directional dominant signals). It is also possible to generate more directional dominant signals than these pairs of signals and related axes. It is possible to achieve this by processing a pair of applied signals in more than one way to generate multiple directional dominant signals in response to a particular pair of applied signals. Before looking at the details of the second phase of the example, it is useful to explain the principle of operation of the second phase.

在已獲得四個方向(L，R，F，B)之每一個的信號強度測量數據後，會要比較在一方向之強度與在相對方向之強度(L比上R，及F比上B)以提供沿著此方向軸之支配的測量數據。由於此例中之四個方向提供彼此成90°的二方向軸 (正交軸)，故此對支配可被解釋為一個二維LR/FB平面上之單一支配向量。此種觀念上或理論上之支配向量可被顯示為第3圖中之例子。雖然此種支配向量在依據本發明之一觀點的一個矩陣或矩陣化程序的作業中為內藏的，但此種支配向量不須外顯地被計算。After the signal strength measurement data of each of the four directions (L, R, F, B) has been obtained, the intensity in one direction and the intensity in the opposite direction (L ratio R, and F ratio B) are compared. ) to provide measurement data that governs the axis along this direction. Since the four directions in this example provide two directions of 90° to each other (orthogonal axis), so the dominance can be interpreted as a single one of the two-dimensional LR/FB planes. Such a conceptual or theoretical dominance vector can be shown as an example in FIG. Although such dominating vectors are built into the operation of a matrix or matrixing program in accordance with one aspect of the present invention, such dominating vectors need not be explicitly calculated.

沿著LR軸之負值可指出朝向左邊之支配，而沿著LR軸之正值可指出朝向右邊之支配。相似地，負的FB值可指出朝向後方之支配，而正的FB值可指出朝向前方之支配。藉由將二個支配值解釋為2D向量之分量，人們可將一信號之支配視覺化成為位於在LR/FB平面上的任何一處。A negative value along the LR axis may indicate a dominance toward the left, while a positive value along the LR axis may indicate a dominance toward the right. Similarly, a negative FB value may indicate a dominance toward the rear, while a positive FB value may indicate a dominance toward the front. By interpreting the two dominant values as components of the 2D vector, one can visualize the dominance of a signal to be anywhere on the LR/FB plane.

在大多數之現代矩陣解碼器(包括Dolby Pro Logic與Dolby Pro Logic II)中，於LR方向之支配係使用L與R之比值來計算，且於FB方向之支配係使用F與B之比值來計算。由於比值與被比較之二信號的量值為獨立的，故其在真實音訊信號之整個自然振幅變異中提供穩定的支配方向。不幸的是，若係用控制數位信號處理器(DSP)之電腦程式來實施，則此種做法需要在程式中之情況述句以選擇分子與分母，以及指定正負號給支配值。更重要的是，諸如在對數定義域中之除法或減法的導出一比值的普通方法需要大量的計算資源。在線性振幅域(如非對數域)中減去此二個數字之較簡單的做法固然在計算上係較有效率，但此種刪減產生之支配信號會隨著信號振幅中之自然變異迅速地變化。In most modern matrix decoders (including Dolby Pro Logic and Dolby Pro Logic II), the LR direction is calculated using the ratio of L to R, and the FB direction is dominated by the ratio of F to B. Calculation. Since the ratio is independent of the magnitude of the second signal being compared, it provides a stable dominant direction throughout the natural amplitude variation of the real audio signal. Unfortunately, if implemented using a computer program that controls a digital signal processor (DSP), this requires a statement in the program to select the numerator and denominator, and a sign to give the dominant value. More importantly, common methods such as deriving a ratio of division or subtraction in a logarithmic domain require a large amount of computational resources. The simpler way of subtracting these two numbers in a linear amplitude domain (such as a non-logarithmic domain) is computationally efficient, but the dominant signal produced by such subtraction will quickly vary with the natural amplitude of the signal amplitude. Change in place.

為降低實施複雜性，本發明之觀點保留比值式比較的大多振幅獨立性，但只需要少很多的計算。To reduce implementation complexity, the present invention retains most of the amplitude independence of the ratio comparison, but requires much less computation.

第二階段之處理器或程序使用線性振幅域的減法器或減法程序來產生多個方向性支配信號，此等減法器或減法程序可獲得每一對被施用之信號的量值間之正或負差。此種減法可用非常低的計算資源來實施。每一個減法之結果被用放大器或放大程序來放大，且被放大之差被施用至一截波器或截波程序，其將每一個被放大之差實質地限制於一正的截波位準與負的截波位準上。或者，放大器或放大程序與截波器或截波程序之順序可使用適當的截波位準來逆轉以產生等效結果。平滑器或平滑程序可對每一個被放大與限制之差作時間平均動作以提供方向性支配信號。The second stage processor or program uses a linear amplitude domain subtractor or subtraction procedure to generate a plurality of directional dominant signals that are positive or between the magnitudes of each pair of applied signals. Negative difference. This subtraction can be implemented with very low computational resources. The result of each subtraction is amplified by an amplifier or amplification procedure, and the amplified difference is applied to a chopper or interceptor that substantially limits each amplified difference to a positive chopping level. With a negative chopping level. Alternatively, the order of the amplifier or amplification procedure and the chopper or interceptor can be reversed using an appropriate chopping level to produce an equivalent result. The smoother or smoothing program can perform a time averaging action on each of the difference between the amplification and the limit to provide a directional dominant signal.

放大器或放大程序之放大因子與截波器或截波程序所在的截波位準間的關係構成量值之正與負臨界值，低於此臨界值的被限制及放大之差可能會實質上具有介於0與該截波位準間的振幅，且高於此臨界值的被限制及放大之差可具有實質上為此截波位準之振幅。雖然此特定轉換函數並非關鍵性的且可能會採取很多形式，但可使此等被限制及放大之差相對於此等差來說在臨界值間實質上為線性的轉換函數係具有非常低之計算需求且為合適的。The relationship between the amplification factor of the amplifier or amplification procedure and the chopping level at which the chopper or chopping program is located constitutes the positive and negative thresholds of the magnitude, and the difference between the limits and amplifications below this threshold may be substantial. There is an amplitude between 0 and the chopping level, and the difference between the limited and the amplified values above the threshold may have an amplitude that is substantially the chopping level for this. Although this particular transfer function is not critical and may take many forms, the difference between such limited and amplified differences is substantially linear with respect to the difference between the critical values. Calculate the requirements and be appropriate.

第二階段之處理器或程序可包括在其處理之際的平滑動作之前或之後對被放大及限制的差的修改，以使導出之方向性支配信號沿著與該方向性支配信號相關的軸被「偏置」。該偏置可為固定的或適應性的。例如，在放大及截波後之一差信號可在振幅上做比例調整，及/或在振幅上做移位(即偏置)，及/或在振幅或符號上以固定方式被限制(例如 作為該振幅、符號、或被放大及限制的差信號之振幅與符號的函數)。其結果可例如包括對非支配信號而不是支配信號施用較少偏置(支配與非支配將在下面做進一步解釋)。對一方向性支配施用「偏置」之例將在下面配合第7圖來描述。The second stage processor or program may include modifications to the amplified and limited differences before or after the smoothing action at the time of processing such that the derived directional dominant signal is along an axis associated with the directional dominant signal Being "offset". This offset can be fixed or adaptive. For example, a difference signal after amplification and chopping may be scaled in amplitude, and/or shifted (ie, offset) in amplitude, and/or fixed in a fixed manner on amplitude or sign (eg, As a function of the amplitude, the sign, or the amplitude and sign of the difference signal that is amplified and limited. The result may, for example, include applying less bias to the non-dominated signal than to the dominant signal (dominance and non-domination will be further explained below). An example of applying a "bias" to a directional control will be described below in conjunction with Figure 7.

在第2圖之第二階段的例子中，施用二對信號L-R與F-B以產生二方向性支配信號LR與FB。如上所述，若有四個中間方向性信號(L，R，F，B)，人們會要藉由比較沿著每一軸之方向性來導出二支配信號分量LR與FB。依據本發明之觀點，此藉由L減R及F減B(或在每一情形中反之亦然)來完成，以提供沿著每一軸的一個振幅差信號。重增益被施用至該等差信號，且被放大之差被截波(硬限制)為-1.0與+1.0。然後被截波之差信號被施用至一時間平滑濾波器。In the example of the second phase of Figure 2, two pairs of signals L-R and F-B are applied to produce bidirectional dominating signals LR and FB. As described above, if there are four intermediate directional signals (L, R, F, B), one would derive the two dominating signal components LR and FB by comparing the directivity along each axis. In accordance with the teachings of the present invention, this is accomplished by subtracting R and F minus B (or vice versa in each case) to provide an amplitude difference signal along each axis. The heavy gain is applied to the equal difference signal, and the amplified difference is chopped (hard limited) to -1.0 and +1.0. The chopped difference signal is then applied to a time smoothing filter.

藉由對該等差信號施用重增益及截波，在一方向上之任何量值的支配基本上便被視為在此方向之一絕對支配。就瞬間方向性由一極性改變為另一極性之信號而言，此操作之結果係類似具有變化頻率與工作週期的長方形波。該時間平滑濾波器將最長方形的波平均掉以提供一連續波，其近似於原始的方向性信號對上彼此之比值。雖然所使用之實際濾波器為設計上之選擇，該濾波器亦可有效地被施作，例如，成為具有約40ms之時間常數的第一階數位IIR低通濾波器。By applying the weight gain and the chopping to the difference signal, the dominance of any magnitude in one direction is essentially considered to be absolutely dominant in one of the directions. In the case of a signal whose instantaneous directivity is changed from one polarity to another, the result of this operation is similar to a rectangular wave having a varying frequency and duty cycle. The temporal smoothing filter averages the most rectangular waves to provide a continuous wave that approximates the ratio of the original directional signal pairs to each other. Although the actual filter used is a design choice, the filter can also be effectively applied, for example, to a first order digital IIR low pass filter having a time constant of about 40 ms.

除了檢測每一軸之支配方向外，呈現「非支配性」可為有利的。例如，純粹向左操縱之輸入信號應在左-右軸上展現強大的支配，但沿著前後軸應絕無支配。另一例為 背景雜訊之極端低位準信號，對此人們偏好不造成任何操縱效果。依照本發明之觀點，如此之一般做法為選擇一臨界值，及對差指定大於-1.0或1.0之臨界值(依該差之符號而定)的量值，及對差指定介於二極值間的小於此臨界值的某量值。有這麼一個可能，即對低於臨界值之所有差值指定0.0的值。為此，在程式控制式DSP中會需要一些情況述句與數值比較。由低複雜度之觀點出發的較佳做法為用大增益來放大差，使得低於臨界值之值的輸出遵循由-1.0至+1.0之線性函數。該增益為該臨界值之倒數。此做法為非常有效率的-增益與截波階段二者可在程式控制式DSP中以DSP之「飽和邏輯」設定(如設定DSP中之一控制暫存器/位元，使得當ALU溢流時，其結果依其正負符號而定地被設定為該平台所呈現之最大正值或最小負值)被施作成算術向左移位(就冪數為2之增益而言)。非冪數為2之增益可以只稍微提高一點處理複雜度來施作。In addition to detecting the formulation of each axis, it may be advantageous to present "non-dominated". For example, an input signal that is manipulated purely to the left should exhibit strong dominance on the left-right axis, but should not be dominant along the front and rear axes. Another example is The extremely low level signal of background noise, people's preference does not cause any manipulation effect. In accordance with the teachings of the present invention, such a general practice is to select a threshold value, and specify a magnitude greater than -1.0 or 1.0 for the difference (depending on the sign of the difference), and specify a difference between the two values for the difference. A certain amount of value less than this threshold. There is a possibility to specify a value of 0.0 for all differences below the threshold. For this reason, some situational statements and numerical comparisons are needed in a program-controlled DSP. A preferred approach from a low complexity point of view is to amplify the difference with a large gain such that the output below the threshold value follows a linear function from -1.0 to +1.0. This gain is the reciprocal of the critical value. This is very efficient - both the gain and the chopping phase can be set in the program-controlled DSP with the "saturation logic" of the DSP (such as setting one of the DSP control registers/bits, so that when the ALU overflows When the result is set to the maximum positive or minimum negative value exhibited by the platform according to its sign, it is applied as an arithmetic left shift (in terms of a gain of 2). A gain with a power of 2 can be applied with only a slight increase in processing complexity.

一個三區域之支配信號(正支配、負支配與非支配)允許在平滑動作前沿著一方向軸之支配與非支配間的分辨。支配與非支配間的分辨促進如上所述之對方向性支配信號適應性施用的「偏置」，其例子在下面配合第7圖給予。如下顯示地，在本發明之觀點中，在平滑動作前分辨來自左環繞操縱信號的獨一左操縱信號及來自右環繞操縱信號的獨一右操縱信號是很有用處的。The dominance signal (positive dominance, negative dominance, and non-domination) of a three region allows the resolution between dominant and non-dominated along a direction axis before smoothing. The discriminating between dominance and non-domination promotes the "offset" of adaptive application of directional dominant signals as described above, an example of which is given below in conjunction with Figure 7. As shown below, in the perspective of the present invention, it is useful to resolve the unique left steering signal from the left surround steering signal and the unique right steering signal from the right surround steering signal prior to the smoothing action.

在本發明之一實務實施例中，為決定來自環繞(左環繞或右環繞)操縱信號之側面(左或右)操縱信號之辨別所必要 的最小增益，解碼了用Dolby Pro Logic II矩陣編碼器所編碼之音樂材料。平均(F-B)差信號就左環繞或右環繞操縱輸入被測量，並且被使用來作為維持左與左環繞(或右與右環繞)間之清楚分辨的最大臨界值(最小增益)之一估計值。在依據本發明之觀點的解碼器的實務實施例中，使用了1024之增益因子，其相當於被常規化為[-1，+1]之信號的約0.001之臨界值。小於0.001之臨界值產生邊緣聽覺改善，而較大的臨界值降低側面(左與右)與環繞(左環繞與右環繞)間之隔離性至不可接受的水準。一般而言，臨界位準不為關鍵性的。In a practical embodiment of the invention, it is necessary to determine the discrimination of the side (left or right) steering signals from the surround (left or right surround) steering signals. The minimum gain is decoded by the music material encoded by the Dolby Pro Logic II matrix encoder. The average (FB) difference signal is measured for the left or right surround manipulation input and is used as an estimate of one of the maximum resolutions (minimum gain) that maintains a clear resolution between left and left surrounds (or right and right surrounds). . In a practical embodiment of a decoder in accordance with the teachings of the present invention, a gain factor of 1024 is used which corresponds to a threshold of about 0.001 that is normalized to a signal of [-1, +1]. A threshold value of less than 0.001 produces edge hearing improvement, while a larger threshold value reduces the isolation between the sides (left and right) and the surround (left and right surround) to an unacceptable level. In general, the critical level is not critical.

為說明此技術，考慮二聲道之立體聲信號，其中左輸入聲道(Lin)為具有0.4尖峰振幅之50Hz的正弦波，且右輸入聲道(Rin)為具有(50*Hz的頻率及1.0的尖峰振幅的正弦波。此等信號在第4圖中被顯示。正弦波之頻率為不相關的，而左聲道之位準為右聲道之位準的0.4倍。使用如上所述之比值式比較，此提供在右方向的0.6的支配(在此被定義為正的)。如在第一階段中所顯示的，L與R中間信號為輸入信號Lin與Rin之量值。To illustrate this technique, consider a two-channel stereo signal in which the left input channel (Lin) is a 50 Hz sine wave with a 0.4 peak amplitude and the right input channel (Rin) is (50*). A sine wave with a frequency of Hz and a peak amplitude of 1.0. These signals are shown in Figure 4. The frequency of the sine wave is irrelevant, while the level of the left channel is 0.4 times the level of the right channel. Using a ratio comparison as described above, this provides a dominance of 0.6 in the right direction (defined herein as positive). As shown in the first stage, the L and R intermediate signals are the magnitudes of the input signals Lin and Rin.

在由R減掉L後，其差被以例如1024來放大(以10位元之算術左移位來實施)，然後在-1.0與+1.0被截波以提供準長方形波。第5圖顯示在截波前與後之差信號。After L is subtracted from R, the difference is amplified by, for example, 1024 (implemented with an arithmetic left shift of 10 bits), then truncated at -1.0 and +1.0 to provide a quasi-rectangular wave. Figure 5 shows the difference signal before and after the chop.

將該準長方形波饋送穿過提供LR方向性支配信號之一平滑濾波器。在輸入信號具有固定位準之例中，方向性支配信號如第6圖中所顯示地最終到達約0.65之值，此接近 使用比值式比較所計算的支配值，且在其附近振盪。該振盪之平滑度為該平滑濾波器的階與特徵之函數。The quasi-rectangular wave is fed through a smoothing filter that provides one of the LR directional dominant signals. In the case where the input signal has a fixed level, the directional dominant signal eventually reaches a value of about 0.65 as shown in Fig. 6, which is close to The dominant value calculated using the ratio comparison is oscillated in the vicinity thereof. The smoothness of the oscillation is a function of the order and characteristics of the smoothing filter.

此例代表在每一個輸入皆具有鉅量無相關的信號之音訊材料，如未編碼的二聲道立體聲信號，其中被截波之放大差的極性被逆轉地非常頻繁。在這些輸入狀況下，該減除/放大/截波所導出之控制信號產生接近比值式比較所獲得的結果。This example represents an audio material that has a large amount of uncorrelated signals at each input, such as an uncoded two-channel stereo signal, where the polarity of the amplified difference of the cut-off is reversed very frequently. Under these input conditions, the control signal derived from the subtraction/amplification/clash produces a result that is close to the ratio comparison.

然而，就在諸如包含於矩陣編碼內容中之單聲道操縱音響源的二聲道中具有共同(即相關)信號的材料而言，被截波之差信號並不包含許多零點跨越。在此類情形中，甚至在或若差信號之極性最終會逆轉時，藉由橫跨到另一極值的平滑調渡，平滑後之控制信號會傾向「鎖定」於二極值(即+1.0與-1.0)之一。此種一支配分量之「鎖定」可被想像為沿著LR/FB平面拉出一個二維支配向量。當二分量均被「鎖定」時，該支配向量被拉至LR/FB平面的四個角落之一。依據本發明之觀點，此種硬式左右平衡藉由提供較離散、單一的輸入聲道至給虛擬器而改善矩陣編碼內容的空間成像。However, just as for materials having a common (i.e., correlated) signal in the two channels of a monophonic sound source included in the matrix encoded content, the chopped difference signal does not contain many zero crossings. In such cases, even if the polarity of the difference signal eventually reverses, the smoothed control signal tends to "lock" to the dipole value by smoothing the transition to another extreme value (ie + One of 1.0 and -1.0). The "lock" of such a component can be imagined as pulling a two-dimensional dominating vector along the LR/FB plane. When both components are "locked", the dominating vector is pulled to one of the four corners of the LR/FB plane. In accordance with the teachings of the present invention, such hard left and right balance improves spatial imaging of matrix encoded content by providing a relatively discrete, single input channel to the virtualizer.

可變增益做法之缺點在於非支配信號可能在被解碼之輸出中漏失。此在大量音響源以很多不同的位準與相位差被混頻在一起的音樂音響源中相當明顯。經常會是，有少數主樂器與人聲在左與右相等地被混頻，儘管還有許多其他較弱的屬音及被加到整體空間與音場周遭的異相位聲 音。由於該解碼器只使用最支配性之音響分量的方向，故對此材料之傳統的可變增益做法會形成幾乎沒有來自後方解碼器輸出的異相位材料之輸出(在此例中為左環繞與右環繞輸出)的結果。A disadvantage of the variable gain approach is that non-dominated signals may be lost in the decoded output. This is quite evident in music sources where a large number of sound sources are mixed together at many different levels and phase differences. It is often the case that there are a few main instruments and vocals that are equally mixed in the left and right, although there are many other weaker tones and are added to the overall space and the sound of the sound field. sound. Since the decoder uses only the direction of the most dominant acoustic component, the traditional variable gain approach to this material results in an output of almost no phase material from the rear decoder output (in this case, left surround and The result of the right surround output).

依據本發明之一觀點，此問題係藉由將FB支配信號朝向後方偏置以確保異相位材料不會從環繞輸出中被移除而獲得緩和。完成此目的之方法為在平滑器濾波前限制FB信號為負值。此在第7圖之例中被顯示。就介於-1.0與1.0間之純長方形波而言，此係相當於用其後有-0.5之固定偏置的平滑濾波器的輸出之一半來按比例縮減。因而，此修改可在平滑器濾波前或後被施加。然而，截波後之差信號可能不是純長方形波。而是，當差信號落在臨界值下而指出沿著特定軸之非支配時，其會包含介於其間的值。當被截波之差信號的量值小於1.0時，限制FB為負值之處理會形成小於在平滑後之可忽略的有效偏置之結果。因而能在此方式平滑前分辨非支配與正或負支配，以允許在雖然給予大部份的其他信號向後的明顯偏置的情況下，左與右操縱信號仍維持與環繞間之高分離度。In accordance with one aspect of the present invention, this problem is mitigated by biasing the FB dominating signal toward the back to ensure that the out-of-phase material is not removed from the surrounding output. The way to accomplish this is to limit the FB signal to a negative value before smoothing the filter. This is shown in the example of Figure 7. For a purely rectangular wave between -1.0 and 1.0, this is equivalent to scaling down one-half the output of a smoothing filter with a fixed offset of -0.5. Thus, this modification can be applied before or after smoother filtering. However, the difference signal after clipping may not be a pure rectangular wave. Rather, when the difference signal falls below a threshold and indicates non-domination along a particular axis, it will contain a value between them. When the magnitude of the chopped difference signal is less than 1.0, the process of limiting the FB to a negative value results in less than a negligible effective offset after smoothing. It is thus possible to resolve the non-dominated and positive or negative dominance before smoothing in this way, allowing the left and right steering signals to maintain a high degree of separation from the surround, while giving a significant bias to the other signals backwards. .

第三階段之處理器或程序產生控制信號，這些控制信號係用於在響應於多個方向性支配信號下，藉由施用一個或多個左右平衡函數(左右平衡函數為代表聲道間之「左右平衡」特徵之轉換函數)至每一個方向性支配信號，來控制適應性矩陣或矩陣化程序。一個或多個左右平衡函數可採 用下列項目之一個或多個：．一個三角轉換函數(如正弦或餘弦轉換函數)，．一個對數轉換函數，．一個線性轉換函數，及．一個三角轉換函數之數學簡化近似。The third stage processor or program generates control signals for applying one or more left and right balance functions in response to the plurality of directional dominant signals (the left and right balance functions are representative of the channels) The left and right balance "characteristic transfer function" to each directional control signal to control the adaptive matrix or matrixing program. One or more left and right balance functions are available Use one or more of the following items: a trigonometric conversion function (such as a sine or cosine conversion function), a logarithmic conversion function, a linear conversion function, and. A mathematically simplified approximation of a trigonometric transfer function.

在此例中，第三階段之目標為取得在前一階段中被計算出的LR與FB支配信號，及導出被施用至被動矩陣之輸出，以產生被解碼之輸出。In this example, the goal of the third stage is to obtain the LR and FB dominating signals that were calculated in the previous stage, and to derive the output that is applied to the passive matrix to produce the decoded output.

對依據本發明之觀點的矩陣解碼器或解碼程序之一般做法是這樣的：在已檢測輸入之某一支配方向性後，強調最接近支配位置的輸出聲道，並解除對最遠支配位置的輸出。在最接近支配位置之二輸出間，其問題可被縮減為成對式左右平衡，其可被表達成一個左右平衡函數。A general approach to a matrix decoder or decoding procedure in accordance with the teachings of the present invention is to emphasize the output channel closest to the dominant position after the dominant directionality of the input has been detected, and to remove the most dominant position. Output. Between the two outputs closest to the dominant position, the problem can be reduced to a pairwise left and right balance, which can be expressed as a left and right balance function.

在二聲道間之最普遍的左右平衡法則為正弦/餘弦法則，其中L=cos(x)*輸入，且R=sin(x)*輸入，而x在0至π/2之間變化。見第8圖。The most common left-right balance rule between two channels is the sine/cosine rule, where L = cos(x)* input, and R = sin(x)* input, and x varies between 0 and π/2. See Figure 8.

每一個解碼器輸出聲道之增益必須以LR與FB函數來表達：LGain=f_L (LR,FB)The gain of each decoder output channel must be expressed in LR and FB functions: LGain=f _L (LR, FB)

RGain=f_R (LR,FB)RGain=f _R (LR, FB)

LsGain=f_Ls (LR,FB)LsGain=f _Ls (LR, FB)

RsGain=f_Ls (LR,FB)RsGain=f _Ls (LR, FB)

可對LR與FB軸施用相同的上述正弦/餘弦左右平衡法則，並獲得第9a與9b圖中顯示之左右平衡曲線，其中panL，panR，panB與panF代表分別來自左、右、後、前之增益分配。The same sine/cosine left and right balance rule can be applied to the LR and FB axes, and the left and right balance curves shown in the figures 9a and 9b are obtained, wherein panL, panR, panB and panF represent the left, right, back, and front respectively. Gain distribution.

了解正弦函數即為相位平移後之餘弦函數後，可只使用餘弦函數來獲得下列的左右平衡公式：panL=cos((LR+1)/2* π/2)After understanding the sine function as the cosine function after phase shifting, you can use only the cosine function to obtain the following left and right balance formula: panL=cos((LR+1)/2* π/2)

panR=sin((LR+1)/2* π/2)=cos((LR-1)/2* π/2)panR=sin((LR+1)/2* π/2)=cos((LR-1)/2* π/2)

panB=cos((FB+1)* π/2)panB=cos((FB+1)* π/2)

panF=sin((FR+1)* π/2)=cos(FR* π/2)panF=sin((FR+1)* π/2)=cos(FR* π/2)

藉由LR/FB平面上之左聲道位置的性質(見第3圖)，LGain應只有在panL與panF二者均為最大值時為最大值，且應隨著支配遠離二軸或其中一軸而減小。此可用panL乘以panF來達成。相同之原理可被施用在RGain，LsGain與RsGain上，且增益之最後公式變成：LGain=panL*panFWith the nature of the left channel position on the LR/FB plane (see Figure 3), LGain should be maximum only when both panL and panF are maximum, and should be spaced away from the two axes or one of the axes. And decrease. This can be achieved by multiplying panL by panF. The same principle can be applied to RGain, LsGain and RsGain, and the final formula of the gain becomes: LGain=panL*panF

RGain=panR*panFRGain=panR*panF

LsGain=panL*panBLsGain=panL*panB

RsGain=panR*panBRsGain=panR*panB

乘法之使用亦可被看作為二正弦/餘弦振幅左右平衡函數之相互比例調整，其中二分量之最小值會成為整體增益可達的最大值。The use of multiplication can also be seen as a mutual scaling of the two sine/cosine amplitude left and right balance functions, where the minimum of the two components becomes the maximum value at which the overall gain can be reached.

第10圖顯示LGain公式之三維呈現，且第11圖顯示所有四個增益相疊之三維呈現Figure 10 shows the 3D rendering of the LGain formula, and Figure 11 shows the 3D rendering of all four gains.

如第8圖所顯示，左右平衡法則係由cos(x)與sin(x)二曲線來合成(sin函數可用cos函數以適當相位平移來取代)。為避免複雜之計算或使用大檢查表，依據本發明之觀點，可替代性地使用介於0與π/2間之餘弦曲線的二階多項式近似。公式y=(1-x² )在0<x<1之範圍內係合理地接近y=(x* π/2)。(見第12圖，其中較低之曲線為近似曲線)。使用此近似所得到之結果可能只會有少到沒有的聽覺差異。As shown in Fig. 8, the left and right balance rule is synthesized by the cos(x) and sin(x) two curves (the sin function can be replaced by the cos function with appropriate phase shift). To avoid complicated calculations or to use large checklists, a second order polynomial approximation of the cosine curve between 0 and π/2 can alternatively be used in accordance with the teachings of the present invention. The formula y=(1-x ² ) is reasonably close to y=(x* π/2) in the range of 0<x<1. (See Figure 12, where the lower curve is an approximate curve). The results obtained with this approximation may only have little to no auditory differences.

由於所預期之音頻輸入源為已被混頻以在L與R間自然地左右平衡二聲道之立體聲，故本發明之一觀點為在計算LGain與RGain時不考慮LR左右平衡分量。在可變增益中之額外左-右平衡在此情況中並不會顯著地改善分離性，原因在於L與R已經被分離得很好了。除了節省一些計算外，其亦藉由避免不必要之增益裝載而允許在前方之較穩定的音場。去除LR分量後，得到這些公式：LGain=panFSince the expected audio input source is stereo that has been mixed to naturally balance the two channels between L and R, one aspect of the present invention is that the left and right balance components of LR are not considered in the calculation of LGain and RGain. The extra left-right balance in the variable gain does not significantly improve the separation in this case because L and R have been separated very well. In addition to saving some calculations, it also allows a more stable sound field in front by avoiding unnecessary gain loading. After removing the LR component, these formulas are obtained: LGain=panF

RGain=panFRGain=panF

LsGain=panL*panBLsGain=panL*panB

RsGain=panR*panBRsGain=panR*panB

這些新的公式之三維呈現在第13圖中顯示。The three-dimensional rendering of these new formulas is shown in Figure 13.

注意，類似之簡化可被施用至Ls增益與Rs增益公式，而不使用額外之LR左右平衡，且使用來源信號內之自然左右平衡以創立二環繞信號間的分離。然而在此情形中，Ls 與Rs之分離被第四階段中發生的被動解碼限制。諸如形成本發明之部分觀點的被動解碼矩陣或矩陣化程序只能達成Ls與Rs間之3dB分離，所以從聲道分離度立場來看，這樣的簡化為不可接受的。為維持較高程度之分離，在LsGain與RsGain之公式中的LR分量被保留。Note that a similar simplification can be applied to the Ls gain and Rs gain equations without using an additional LR left and right balance, and using the natural left and right balance within the source signal to create separation between the two surround signals. In this case, however, Ls The separation from Rs is limited by the passive decoding that occurs in the fourth phase. A passive decoding matrix or matrixing procedure such as forming part of the present invention can only achieve a 3 dB separation between Ls and Rs, so such simplification is unacceptable from a channel separation standpoint. In order to maintain a high degree of separation, the LR component in the formula of LsGain and RsGain is retained.

將每一個左右平衡項中之餘弦代入多項式近似，可為每一個增益因子導出最後公式：LGain=1-FB² Substituting the cosine of each left and right balance term into a polynomial approximation, the final formula can be derived for each gain factor: LGain=1-FB ²

當FB=0→LGain=1When FB=0→LGain=1

當FB=-1→LGain=0When FB=-1→LGain=0

RGain=1-FB² RGain=1-FB ²

當FB=0→RGain=1When FB=0→RGain=1

當FB=-1→RGain=0When FB=-1→RGain=0

LsGain=[1-((LR+1)/2)² ]*[1-(FB+1)² ]LsGain=[1-((LR+1)/2) ² ]*[1-(FB+1) ² ]

當FB=0→LsGain=0When FB=0→LsGain=0

當FB=-1及LR=-1→LsGain=1When FB=-1 and LR=-1→LsGain=1

當FB=-1及LR=1→LsGain=0When FB=-1 and LR=1→LsGain=0

RsGain=[1-((LR-1)/2)² ]*[1-(FB+1)² ]RsGain=[1-((LR-1)/2) ² ]*[1-(FB+1) ² ]

當FB=0→RsGain=0When FB=0→RsGain=0

當FB=-1及LR=-1→RsGain=0When FB=-1 and LR=-1→RsGain=0

當FB=-1及LR=1→RsGain=1When FB=-1 and LR=1→RsGain=1

參照第14圖，控制信號LGain、RGain、LsGain及RsGain係由施用左右平衡函數至一方向性支配信號、及/或施用一 左右平衡函數至一方向性支配信號、與施用一左右平衡函數另一方向性支配信號之乘積來導出，其中各個左右平衡函數係可與其他左右平衡函數之全部或部分不同。該等左右平衡函數並非為在此等n個輸入音訊信號中所固有的平衡函數。在此例中，此等方向軸之一為一個左右軸，且該等左右平衡函數為不包括左右平衡分量的左右平衡函數。下列項目可應用於此實施例：LR方向性支配信號被施用至一panL左右平衡函數與一panR函數；FB方向性支配信號(抑如第2圖中之無偏置或如第7圖中之有偏置)被施用至一panF左右平衡函數與一panB左右平衡函數；施用panF函數至FB方向性支配信號之結果被如同LGain與Rgain一般地施用至第四階段之被動解碼器或解碼程序；施用panB函數至FB支配信號之結果被乘以施用panL函數至LR支配信號之結果，且被如同LsGain一般地施用至第四階段被動解碼器或解碼程序；施用panR函數至LR支配信號之結果被乘以施用panB函數至FB支配信號之結果，且被如同RsGain一般地施用至第四階段被動解碼器或解碼程序。Referring to Figure 14, the control signals LGain, RGain, LsGain, and RsGain are applied by applying a left and right balance function to a directional dominant signal, and/or applying one The left and right balance functions are derived from the product of a directional control signal and a directional control signal applied to the left and right balance functions, wherein each of the left and right balance functions may be different from all or part of the other left and right balance functions. These left and right balance functions are not the balance functions inherent in the n input audio signals. In this example, one of the directional axes is a left and right axis, and the left and right balance functions are left and right balance functions that do not include left and right balance components. The following items can be applied to this embodiment: the LR directional dominant signal is applied to a panL left and right balance function and a panR function; the FB directional dominant signal (as in Figure 2, there is no offset or as in Figure 7) The offset is applied to a panF balance function and a panB balance function; the result of applying the panF function to the FB directional control signal is applied to the passive decoder or decoding program of the fourth stage as generally as LGain and Rgain; The result of applying the panB function to the FB dominating signal is multiplied by the result of applying the panL function to the LR dominating signal, and is generally applied to the fourth stage passive decoder or decoding program as LsGain; the result of applying the panR function to the LR dominating signal is Multiply the result of applying the panB function to the FB dominating signal and is generally applied to the fourth stage passive decoder or decoding procedure as RsGain.

第15圖顯示響應於m個音訊信號而產生n個音訊信號之一被動矩陣或矩陣化程序，與振幅比例調整器或振幅比例調整程序，其各在響應於時間變化振幅比例調整因子控制信號下將被動矩陣或矩陣化程序所產生的音訊信號之一作振幅比例調整，以產生n個音訊輸出信號，其中多個時間變化控制信號為n個時間變化振幅比例調整因子控制信 號，即用於將被動矩陣或矩陣化程序所產生之每一個音訊信號作振幅比例調整者。在第14圖之例中，其有二個音訊輸出信號Lin與Rin、四個音訊輸出信號LOut，ROut，LsOut，RsOut，及四個比例調整因子控制信號LGain，RGain，LsGain與RsGain(來自第三階段)。Figure 15 shows a passive matrix or matrixing procedure for generating n audio signals in response to m audio signals, and an amplitude scaler or amplitude scaling procedure, each in response to a time varying amplitude scale adjustment factor control signal One of the audio signals generated by the passive matrix or the matrixing process is amplitude-scaled to generate n audio output signals, wherein the plurality of time-varying control signals are n time-varying amplitude-proportional adjustment factor control signals No. is used to adjust the amplitude ratio of each audio signal generated by the passive matrix or matrixing program. In the example of Fig. 14, there are two audio output signals Lin and Rin, four audio output signals LOut, ROut, LsOut, RsOut, and four proportional adjustment factor control signals LGain, RGain, LsGain and RsGain (from the first Three stages).

在第15圖之例中，四個矩陣或矩陣化程序可用下列公式來特徵化：LOut=LGain*(a*Lin+b*Rin)In the example of Figure 15, the four matrix or matrixing procedures can be characterized by the following formula: LOut=LGain*(a*Lin+b*Rin)

ROut=RGain*(c*Lin+d*Rin)ROut=RGain*(c*Lin+d*Rin)

LsOut=LsGain*(e*Lin+f*Rin)LsOut=LsGain*(e*Lin+f*Rin)

RsOut=RsGain*(g*Lin+h*Lin)RsOut=RsGain*(g*Lin+h*Lin)

此處如第15圖所示之a至h為矩陣係數。該等係數a至h可被選用以媒配在Dolby Pro Logic II編碼/解碼系統中所使用者，此處：a=1.0,b=0.0,c=0.0,d=1.0,e=0.8710,f=-0.4898,g=0.4898,h=0.8710Here, a to h shown in Fig. 15 are matrix coefficients. These coefficients a to h can be selected to match the user in the Dolby Pro Logic II encoding/decoding system, where: a=1.0, b=0.0, c=0.0, d=1.0, e=0.8710,f =-0.4898, g=0.4898, h=0.8710

此提供了下列最後公式：LOut=LGain*LinThis provides the following final formula: LOut=LGain*Lin

ROut=RGain*RinROut=RGain*Rin

LsOut=LsGain*(0.8710*Lin-0.4898*Rin)LsOut=LsGain*(0.8710*Lin-0.4898*Rin)

RsOut=RsGain*(0.8710*Rin-0.4898*Lin)RsOut=RsGain*(0.8710*Rin-0.4898*Lin)

第16圖顯示此範例之所有四個階段的縱覽，並指出其 相互關係。Figure 16 shows an overview of all four phases of this example and points out Interrelationship.

本發明可用硬體或軟體或二者之組合(如可程設的邏輯陣列)來實施。除非特別明定，否則被包括做為本發明之部分的法則並非固定地與任何特定電腦或其他裝置相關。特別是，諸如數位信號處理器之多種一般用途機器可配合依照此處所教授而寫成的程式來使用，或可較便利地構建更專業之裝置(如積體電路)以執行所要求的方法步驟。因而，本發明可在執行一個或多個程式化電腦系統的一個或多個電腦程式中被實施，各可程式化電腦系統包含至少一個處理器、至少一個個資料儲存系統(包括依電性與非依電性記憶體及/或儲存元件)、至少一個輸入裝置或輸入埠、及至少一個輸出裝置或輸出埠。程式碼被施用在輸入資料上以執行如此處所描述的函數並產生輸出資訊。該輸出資訊以習知方式被施用至一個或多個輸出裝置。The invention may be implemented in hardware or software or a combination of both, such as a programmable logic array. Unless specifically stated otherwise, the rules included as part of the invention are not fixedly related to any particular computer or other device. In particular, a variety of general purpose machines, such as digital signal processors, can be used in conjunction with programs written in accordance with the teachings herein, or more convenient to construct more specialized devices (e.g., integrated circuits) to perform the required method steps. Thus, the present invention can be implemented in one or more computer programs executing one or more stylized computer systems, each programmable computer system including at least one processor, at least one data storage system (including power and Non-electrical memory and/or storage element), at least one input device or input port, and at least one output device or output port. The code is applied to the input data to perform the functions as described herein and to generate output information. The output information is applied to one or more output devices in a conventional manner.

每一個此等程式可用任何所欲之電腦語言(包括機器語言、組合語言、或高階程序語言、邏輯語言、或目標導向程式語言)以與電腦系統通訊。在任何情形中，該語言可為編譯與解譯語言。Each of these programs can communicate with a computer system in any desired computer language (including machine language, combination language, or higher level programming language, logical language, or target oriented programming language). In any case, the language can be a compiled and interpreted language.

每一個此等電腦程式係較佳地被儲存或下載至通用或特殊目的之可程式化的電腦可讀取儲存媒體或裝置中(如固態記憶體或媒體，或磁性或光學媒體)，以在儲存媒體或裝置被電腦系統讀取時組配及操作電腦來執行此處所描述之程序。亦可考慮將本發明系統以用電腦程式來組配的電 腦可讀取儲存媒體來實施，其中如此組配之儲存媒體致使電腦系統以特定與預先定義的方式操作，以執行此處所描述之函數。Each such computer program is preferably stored or downloaded to a general purpose or special purpose programmable computer readable storage medium or device (such as solid state memory or media, or magnetic or optical media) for The storage media or device is assembled and operated by a computer system to perform the procedures described herein. It is also conceivable to use the computer of the present invention in a computer program. The brain is readable by a storage medium, wherein the storage medium so configured causes the computer system to operate in a specific and predefined manner to perform the functions described herein.

在適於控制數位信號處理器之電腦程式中被實施的本發明之實務實施例已有使用30行的C語言程式碼來實施，以大概是3MIPS來運轉，並在虛擬上未使用記憶體。這大概是Dolby Pro Logic II解碼器所預估的使用的MIPS的15%。程序可整個留在時域中且以逐一樣本之基準來執行(無區塊程序)。為了使每一個樣本之執行時間最小化，施作可避免使用分支與諸如平方根、正弦、餘弦與除法等之數學函數。施作亦可避免使用檢查表與先行延遲，其提高記憶體需求且增加執行時間。本發明之觀點可用非常簡單之電腦程式與非常基本之數位信號處理器來實施。特別是從簡單性來看，本發明之觀點亦可使用類比電路來實施。The practical embodiment of the present invention implemented in a computer program suitable for controlling a digital signal processor has been implemented using 30 lines of C language code, operating at approximately 3 MIPS, and memory is not used virtually. This is probably 15% of the MIPS used by the Dolby Pro Logic II decoder. The program can be left entirely in the time domain and executed on a benchmark basis (no block program). In order to minimize the execution time of each sample, the application can avoid the use of branches and mathematical functions such as square root, sine, cosine and division. The implementation also avoids the use of checklists and advance delays, which increase memory requirements and increase execution time. The perspective of the present invention can be implemented with a very simple computer program and a very basic digital signal processor. In particular, from the standpoint of simplicity, the idea of the present invention can also be implemented using analog circuits.

本發明之多個實施例已被描述。不過，將了解各種修改可被做成而不偏離本發明之精神與領域。例如，此處所描述之一些步驟在順序上可為獨立的，且因而可以與此處所描述之不同之順序來執行。Various embodiments of the invention have been described. However, it will be appreciated that various modifications may be made without departing from the spirit and scope of the invention. For example, some of the steps described herein can be independent in sequence and thus can be performed in a different order than described herein.

Lin‧‧‧左輸入信號Lin‧‧‧ left input signal

Rin‧‧‧右輸入信號Rin‧‧‧Right input signal

FB、LR‧‧‧方向性支配信號FB, LR‧‧‧ directional dominant signal

L、R、F、B‧‧‧中間控制信號L, R, F, B‧‧‧ intermediate control signals

panL‧‧‧左增益分配panL‧‧‧Left gain allocation

panR‧‧‧右增益分配panR‧‧‧right gain distribution

panB‧‧‧後增益分配panB‧‧‧ post gain allocation

panF‧‧‧前增益分配panF‧‧‧ front gain distribution

Lgain、Rgain、LsGain、RsGain‧‧‧控制信號Lgain, Rgain, LsGain, RsGain‧‧‧ control signals

a~h‧‧‧矩陣係數a~h‧‧‧matrix coefficient

Lout‧‧‧左輸出信號Lout‧‧‧left output signal

Rout‧‧‧右輸出信號Rout‧‧‧Right output signal

LsOut‧‧‧左環繞輸出信號LsOut‧‧‧ left surround output signal

RsOut‧‧‧右環繞輸出信號RsOut‧‧‧Right surround output signal

第1圖為示意性功能方塊圖，顯示依據本發明之觀點用於由多個音訊輸入信號導出多對中間控制信號的處理器或程序之一例。1 is a schematic functional block diagram showing an example of a processor or program for deriving a plurality of pairs of intermediate control signals from a plurality of audio input signals in accordance with the teachings of the present invention.

第2圖為示意性功能方塊圖，顯示依據本發明之觀點用於導出多個方向性支配信號之處理器或程序的一例。Figure 2 is a schematic functional block diagram showing an example of a processor or program for deriving a plurality of directional dominant signals in accordance with the teachings of the present invention.

第3圖顯示以正交的LR與FB軸為基礎之二維平面中的觀念性或理論性方向支配向量的一例。Figure 3 shows an example of a conceptual or theoretical directional dominating vector in a two-dimensional plane based on orthogonal LR and FB axes.

第4圖為信號振幅對上時間之理想化圖示，其分別顯示一個二聲道立體聲信號的絕對值L與R。Figure 4 is an idealized representation of the signal amplitude versus time, which shows the absolute values L and R of a two-channel stereo signal, respectively.

第5圖為信號振幅對上時間之理想化圖示，顯示由R減掉L，以及乘上其差然後在-1.0與1.0截波以提供一準長方形波二者之結果。Figure 5 is an idealized representation of the signal amplitude versus time, showing the result of subtracting L from R and multiplying the difference and then truncating at -1.0 and 1.0 to provide a quasi-rectangular wave.

第6圖為信號振幅對上時間之理想化圖示，顯示由將該準長方形波饋送穿過一平滑器濾波器所致之平滑後的LR中間控制信號。Figure 6 is an idealized representation of the signal amplitude versus time, showing the smoothed LR intermediate control signal resulting from feeding the quasi-rectangular wave through a smoother filter.

第7圖為示意式功能方塊圖，顯示依據第2圖顯示之本發明的觀點之處理器或程序的修正。Figure 7 is a schematic functional block diagram showing the modification of the processor or program in accordance with the teachings of the present invention shown in Figure 2.

第8圖為以弧線表示的增益對上角度之理想化圖示，顯示左(L)與右(R)音頻聲道間之共同左右平衡法則(pan-law)，即正弦/餘弦左右平衡法則。Figure 8 is an idealized representation of the gain versus upper angle in arcs, showing the common left-right balance rule (pan-law) between the left (L) and right (R) audio channels, ie the sine/cosine left and right balance rule .

第9a圖為當第8圖中之相同的正弦/餘弦左右平衡法則被施用至LR軸時，分別就panL與panR顯示增益對上方向性支配信號位準之理想化圖示。Figure 9a shows an idealized representation of the gain versus the directional dominant signal level for panL and panR, respectively, when the same sine/cosine left and right balance rule is applied to the LR axis in Figure 8.

第9b圖為當第8圖中之相同的正弦/餘弦左右平衡法則被施用至FB軸時，分別就panB與panF顯示增益對上方向性支配信號位準之理想化圖示。Figure 9b shows an idealized representation of the gain versus the directional dominant signal level for panB and panF, respectively, when the same sine/cosine left and right balance law is applied to the FB axis in Figure 8.

第10圖為一理想化圖示，顯示LGain公式之準三維呈現(其三軸為正規化之增益及FB與LR之值)。Figure 10 is an idealized diagram showing the quasi-three-dimensional representation of the LGain formula (its three axes are the normalized gain and the values of FB and LR).

第11圖為一理想化圖示，顯示LGain，RGain，LsGain 與RsGain公式之準三維呈現(其三軸為正規化之增益及FB與LR之值)。Figure 11 is an idealized diagram showing LGain, RGain, LsGain Quasi-three-dimensional representation with the RsGain formula (its three axes are the normalized gain and the values of FB and LR).

第12圖為一理想化圖示，顯示一餘弦波與介於0及π/2間之餘弦的第二階多項式近似。Figure 12 is an idealized diagram showing a cosine wave approximation of a second order polynomial with a cosine between 0 and π/2.

第13圖為一理想化圖示，顯示LGain，RGain，LsGain與RsGain公式之準三維呈現(其三軸為正規化之增益及FB與LR之值)。Figure 13 is an idealized diagram showing the quasi-three-dimensional representation of the LGain, RGain, LsGain and RsGain formulas (the three axes are the normalized gain and the values of FB and LR).

第14圖為示意式功能方塊圖，顯示依據本發明的之觀點，用於由多個方向性支配信號導出多個控制信號的處理器或程序之一例。Figure 14 is a schematic functional block diagram showing an example of a processor or program for deriving a plurality of control signals from a plurality of directional dominant signals in accordance with the teachings of the present invention.

第15圖為示意式功能方塊圖，顯示依據本發明的之觀點，用於由音訊輸入信號與多個控制信號導出多個音訊輸出信號的適應性矩陣或矩陣化程序之一例。Figure 15 is a schematic functional block diagram showing an example of an adaptive matrix or matrixing procedure for deriving a plurality of audio output signals from an audio input signal and a plurality of control signals in accordance with the teachings of the present invention.

第16圖為一示意式方塊圖，顯示本例全部四個階段的綜述，並指出其相互關係。Figure 16 is a schematic block diagram showing an overview of all four phases of this example and indicating their relationship.

Lin‧‧‧左輸入信號Lin‧‧‧ left input signal

Rin‧‧‧右輸入信號Rin‧‧‧Right input signal

FB、LR‧‧‧方向性支配信號FB, LR‧‧‧ directional dominant signal

L、R、F、B‧‧‧中間控制信號L, R, F, B‧‧‧ intermediate control signals

一種用於處理音訊信號之方法，其包含下列步驟：由m個音訊輸入信號導出n個音訊輸出信號，其中m與n為正整數，並且該等n個音訊輸出信號係使用響應於n個時變控制信號的一個適應性矩陣或矩陣化程序來導出，該矩陣或矩陣化程序響應於m個音訊信號而產生n個音訊信號；由該等m個音訊輸入信號來導出該等n個控制信號，該導出步驟係使用：一個被動矩陣或矩陣化程序，該被動矩陣或矩陣化程序響應於該等m個音訊輸入信號而產生數對信號，其中有表示沿著一第一方向軸之相對向方向信號強度的第一對信號，以及表示沿著一第二方向軸之相對向方向信號強度的第二對信號；一個第一處理器或程序，該第一處理器或程序響應於該等輸入信號對而產生多個方向性支配信號，其中至少一個方向性支配信號係與一個第一方向軸有關，並且至少一個其他方向性支配信號係與一個第二方向軸有關；以及一個第二處理器或程序，該第二處理器或程序響應於該等方向性支配信號而產生該等控制信號，其特徵在於：產生多個方向性支配信號之該第一處理器或程序使用：可獲得每一對信號之量值間的一個正差或負差的 數個線性振幅域減除器或減除程序；將每一個該等差實質上限制於一個正截波位準上與一個負截波位準上的一個截波器或截波程序；施行(1)放大各個該等差以使該截波器或截波程序限制各個該等被放大的差，或(2)放大各個該等被限制的差，之動作的一個放大器或放大程序；以及將每一個被放大及限制之差或被限制及放大之差作時間平均動作的一個平滑器或平滑程序。 A method for processing an audio signal, comprising the steps of: deriving n audio output signals from m audio input signals, wherein m and n are positive integers, and the n audio output signals are used in response to n Deriving an adaptive matrix or matrixing procedure of the variable control signal, the matrix or matrixing process generating n audio signals in response to the m audio signals; deriving the n control signals from the m audio input signals The deriving step uses: a passive matrix or matrixing program that generates pairs of signals in response to the m audio input signals, wherein the relative directions along a first direction axis are present a first pair of signals of direction signal strength, and a second pair of signals indicative of relative direction signal strength along a second direction axis; a first processor or program responsive to the inputs The signal pair produces a plurality of directional dominant signals, wherein at least one directional dominant signal is associated with a first directional axis and at least one other directional branch The signal system is associated with a second direction axis; and a second processor or program that generates the control signals in response to the directional dominant signals, characterized by: generating a plurality of directivities The first processor or program that governs the signal: a positive or negative difference between the magnitudes of each pair of signals is obtained a plurality of linear amplitude domain subtractors or subtraction programs; each of the equalities is substantially limited to a chopper or a chopping program at a positive chopping level and a negative chopping level; 1) amplifying each of the equalities such that the chopper or chopping program limits each of the amplified differences, or (2) amplifying each of the limited differences, an amplifier or amplification procedure; Each smoothing or smoothing procedure that is amplifying and limiting the difference or the difference between the limited and amplified times as a time-averaging action. 如申請專利範圍第1項所述之方法，其中對於不相關的數個音訊輸入信號而言，各個方向性支配信號係根據信號對的一個比值來近似一個方向性支配信號，而對於相關的數個音訊輸入信號而言，該方向性支配信號傾向負截波位準或正截波位準。 The method of claim 1, wherein for the unrelated plurality of audio input signals, each directional dominant signal approximates a directional dominant signal according to a ratio of the signal pairs, and for the related number For an audio input signal, the directional dominant signal tends to a negative or positive chopping level. 如申請專利範圍第2項所述之方法，其中高於該正截波位準的一個差指出沿著一個方向軸的一個正支配，而低於該負截波位準的一個差指出沿著一個方向軸的一個負支配，並且介於該正截波位準與該負截波位準間的一個差指出沿著一個方向軸的非支配。 The method of claim 2, wherein a difference above the positive chopping level indicates a positive dominance along an axis of direction, and a difference below the negative chopping level indicates a A negative control of one of the directional axes, and a difference between the positive chopping level and the negative chopping level indicates a non-dominance along one direction axis. 如申請專利範圍第3項所述之方法，其中產生多個方向性支配信號之該處理器或程序在沿著一個方向軸上有非支配時，與有正支配或負支配時不同地，修改該被放大及限制的差或該被限制及放大的差。 The method of claim 3, wherein the processor or program that generates the plurality of directional dominant signals is modified when it is non-dominated along one direction axis, and is modified differently when it is dominant or negatively dominated. The difference that is amplified and limited or the difference that is limited and amplified. 如申請專利範圍第1項所述之方法，其中產生多個方向性支配信號的該處理器或程序，亦限制在一平滑器或平滑程序之前之一截波器或截波程序之輸出的正量值或 負量值。 The method of claim 1, wherein the processor or program that generates the plurality of directional dominant signals is also limited to the output of one of the chopper or the chopping program before a smoother or smoothing procedure. Measured value or Negative value. 如申請專利範圍第5項所述之方法，其中產生多個方向性支配信號的該處理器或程序，限制在該平滑器或平滑程序之前之至少一個該等截波器或截波程序的該輸出之該正量值。 The method of claim 5, wherein the processor or program that generates the plurality of directional dominant signals limits the at least one of the choppers or interceptors prior to the smoother or smoothing procedure The positive value of the output. 如申請專利範圍第6項所述之方法，其中該第一方向軸為一前/後軸，並且產生多個方向性支配信號的該處理器或程序限制處理一個前/後軸方向性支配信號之該截波器或截波程序的該輸出之該正量值。 The method of claim 6, wherein the first direction axis is a front/rear axis, and the processor or program that generates the plurality of directional dominant signals limits processing of a front/rear axis directional dominant signal The positive value of the output of the chopper or chopping program. 如申請專利範圍第1項所述之方法，其中響應於該等多個方向性支配信號而產生該等控制信號的該第二處理器或程序，對各個該等多個方向性支配信號施用至少一個左右平衡函數。 The method of claim 1, wherein the second processor or program that generates the control signals in response to the plurality of directional dominant signals applies at least each of the plurality of directional dominant signals A left and right balance function. 如申請專利範圍第8項所述之方法，其中一個或多個該等左右平衡函數採用一個三角轉換函數。 The method of claim 8, wherein one or more of the left and right balance functions employ a triangular transfer function. 如申請專利範圍第8項所述之方法，其中一個或多個該等左右平衡函數採用一個對數轉換函數。 The method of claim 8, wherein one or more of the left and right balance functions employ a logarithmic transfer function. 如申請專利範圍第8項所述之方法，其中一個或多個該等左右平衡函數採用一個線性轉換函數。 The method of claim 8, wherein one or more of the left and right balance functions employ a linear transfer function. 如申請專利範圍第8項所述之方法，其中一個或多個該等左右平衡函數採用一個三角轉換函數的一個數學簡化近似。 The method of claim 8, wherein the one or more of the left and right balance functions employ a mathematically simplified approximation of a triangular transfer function. 如申請專利範圍第8項所述之方法，其中該等控制信號係以下列項目來導出： 將一個左右平衡函數在一個方向性支配信號上的施用結果，及/或將一個左右平衡函數在一個方向性支配信號上的施用結果以及將一個左右平衡函數在另一個方向性支配信號上的施用結果之乘積，其中各個左右平衡函數可為與其他所有的左右平衡函數不同或與其他左右平衡函數中之一些不同。 The method of claim 8, wherein the control signals are derived by the following items: Application of the application of a left and right balance function to a directional dominant signal, and/or application of a left and right balance function to a directional dominant signal and application of a left and right balance function to another directional dominant signal The product of the results, wherein each of the left and right balance functions may be different from all other left and right balance functions or different from some of the other left and right balance functions. 如申請專利範圍第8項所述之方法，其中該等左右平衡函數並非在該等n個音訊輸入信號中所固有的左右平衡函數。 The method of claim 8, wherein the left and right balance functions are not left and right balance functions inherent in the n audio input signals. 如申請專利範圍第14項所述之方法，其中該等方向軸之一為一個左/右軸，並且該等左右平衡函數為不包括一個左/右側左右平衡分量的左右平衡函數。 The method of claim 14, wherein one of the direction axes is a left/right axis, and the left and right balance functions are left and right balance functions that do not include a left/right left and right balance component. 如申請專利範圍第8項所述之方法，其中該等n個時變比例調整因子信號中之至少一些信號係由一個單一左右平衡函數在一個方向性支配信號上的施用結果來導出，並且該等n個時變比例調整因子信號中之其他信號係由一個單一左右平衡函數在一個方向性支配信號上的施用結果與另一個單一左右平衡函數在另一個方向性支配信號上的施用結果之乘積來導出。 The method of claim 8, wherein at least some of the n time-varying proportional adjustment factor signals are derived from a result of application of a single left-right balance function on a directional dominant signal, and The other signals in the n time-varying proportional adjustment factor signals are the product of the application result of a single left-right balance function on one directional dominant signal and the application result of another single left-right balance function on the other directional dominant signal. To export. 如申請專利範圍第16項所述之方法，其中該等方向性支配信號中的一個支配信號之該方向軸為一個左/右軸，並且該等方向性支配信號中的另一個支配信號的該方向軸為一個前/後軸，其中至少一些該等n個時變比例調 整因子信號係由將一個單一左右平衡函數在前/後方向性支配信號上的施用結果來導出，並且至少一些該等n個時變比例調整因子信號係由一個單一左右平衡函數在左/右方向性支配信號上的施用結果與另一個單一左右平衡函數在該前/後方向性支配信號上的施用結果之乘積來導出。 The method of claim 16, wherein the direction axis of one of the directional dominant signals is a left/right axis, and the other of the directional signals governs the signal The direction axis is a front/rear axis, at least some of the n time-varying proportional adjustments The factorial signal is derived from the application of a single left and right balance function on the anterior/posterior directional dominant signal, and at least some of the n time varying proportional adjustment factor signals are left/right by a single left and right balance function The result of the application on the directional dominant signal is derived from the product of the application of another single left and right balance function on the anterior/posterior directional dominating signal. 如申請專利範圍第1項所述之方法，進一步包含由該等n個音訊輸出信號來導出p個音訊信號之步驟，其中p為二，並且該等p個音訊信號係使用一個虛擬器或虛擬化程序從該等n個音訊信號來導出的，使得當該等p個音訊信號被施用在一對換能器上時，相對該等換能器適當地被定位的一個聆聽者感覺該等n個音訊信號彷彿來自與該等換能器之所在不同的位置。 The method of claim 1, further comprising the step of deriving p audio signals from the n audio output signals, wherein p is two, and the p audio signals are using a virtual device or virtual The program is derived from the n audio signals such that when the p audio signals are applied to a pair of transducers, a listener positioned appropriately with respect to the transducers senses the n The audio signals appear to be from a different location than the transducers. 如申請專利範圍第18項所述之方法，其中該虛擬器或虛擬化程序包括一個或多個頭部相關轉換函數在該等n個音訊輸出信號中之一些輸出信號上的施用。 The method of claim 18, wherein the virtualizer or virtualization program includes the application of one or more head related conversion functions on some of the n audio output signals. 如申請專利範圍第18項所述之方法，其中該對換能器為一對耳機。 The method of claim 18, wherein the pair of transducers is a pair of earphones. 如申請專利範圍第18項所述之方法，其中該對換能器為一對喇叭。 The method of claim 18, wherein the pair of transducers are a pair of speakers. 一種適於執行如申請專利範圍第1至21項任一項所述之方法的裝置。 An apparatus adapted to perform the method of any one of claims 1 to 21. 一種儲存於電腦可讀媒體上的電腦程式，該程式係用以使一電腦執行如申請專利範圍第1至21項任一項所述之 方法。 A computer program stored on a computer readable medium for causing a computer to perform the method of any one of claims 1 to 21 method. 如申請專利範圍第1至21項任一項所述之方法，其中m為2，且n為4或5。 The method of any one of claims 1 to 21, wherein m is 2 and n is 4 or 5.

• 2006
  • 2006-10-23 TW TW95138970A patent/TWI420918B/en active
  • 2006-11-16 EP EP06837740A patent/EP1964443B1/en active Active
  • 2006-11-16 WO PCT/US2006/044447 patent/WO2007067320A2/en active Application Filing
  • 2006-11-16 CN CN2006800519731A patent/CN101336563B/en active Active
• 2009
  • 2009-02-10 HK HK09101179.2A patent/HK1123663A1/en unknown

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