Method and apparatus to create a sound field
Abstract
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Expired 27 March 2022, 4.5 years ago.
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21 claims: 20 independent, 1 dependent
- 1出力変換器のアレイを用いて、 異なるそれぞれの方向に送出される 複数のサウンド・チャネルから成る音場を作り出す方法であって、 各チャネル毎に、各出力変換器に関して第1遅延値を選択するステップであって、前記第1遅延値を、前記それぞれの変換器の前記アレイにおける位置 と前記チャネルが送出される方向 に応じて選択する、ステップと、 各チャネル毎に第2遅延値を選択するステップであって、前記第2遅延値を、前記アレイから聴取者までの、当該チャネルの音波の予想伝搬距離に応じて選択する、ステップと、 各出力変換器に関して、各チャネルを表す信号を遅延させた複製を得るステップであって、各遅延複製を、前記第1遅延値から成る第1成分と前記第2遅延値から成る第2成分とを有する値だけ遅延させる、ステップと、 各出力変換器に関して、前記遅延複製の各々を互いに加算するステップと、 から成り、 聴取者に到達する前に、少なくとも1つのチャネルの音波を壁のような表面で跳ね返す、前記音場を作り出す 方法。
- 2請求項 1 記載の方法において、前記チャネルを表す各信号を複製する前に、前記信号に前記第2遅延を与え、次いで前記それぞれの第1遅延値だけ各複製を遅延させる、方法。
- 3請求項1または2 記載の方法において、前記第2遅延値の選択を、全サウンド・チャネルの対応する部分が実質的に同時に前記聴取者に到達するように行う、方法。
- 4異なるそれぞれの方向に送出される複数のサウンド・チャネルから成る 音場を作り出す装置であって、 異なるサウンド・チャネルを表す複数のそれぞれの信号のための複数の入力と、 出力変換器のアレイと、 各出力変換器に関して、各それぞれの入力信号の複製を得るように構成された複製手段と、 前記それぞれの出力変換器の前記アレイ内における位置 と前記チャネルが送出される方向 に応じて選択したそれぞれの第1遅延値だけ、各信号の各複製を遅延するように構成された第1遅延手段と、 前記アレイから聴取者までの当該チャネルの音波の予想伝搬距離に応じて、各チャネル毎に選択した第2遅延値だけ、各信号の各複製を遅延させるように構成された第2遅延手段と、 各出力変換器毎に設けた加算手段であって、各信号の2度遅延した前記複製を互いに加算して各出力変換器への出力信号を生成する加算手段と、 から成り、 前記少なくとも1つのチャネルの音波の予想伝播距離は、壁のような表面での跳ね返りを含む、前記音場を作り出す 装置。
- 5請求項 4 記載の装置において、前記第2遅延手段が、前記複製手段によって複製が作成される前に、前記入力信号を遅延させるように構成されている、装置。
- 6請求項 4または5 に記載の装置において、前記第2遅延手段が、全サウンド・チャネルが実質的に同時に聴取者に到達するように、各チャネル毎に前記第2遅延を選択するように構成されている、装置。
- 7中央チャネルと少なくとも1つのサラウンド・サウンド・チャネルから成る音場を作り出し、出力変換器のアレイを用いて、前記少なくとも1つのサラウンド・サウンド・チャネルを所定の方向に送出する方法であって、 前記少なくとも1つのサラウンド・サウンド・チャネルに対して、各出力変換器に関して第1遅延値を選択するステップであって、前記第1遅延値を、前記それぞれの変換器の前記アレイにおける位置に応じて選択し、前記チャネルを前記所定の方向に送出する、ステップと、 前記中央チャネルに対して第2遅延値を選択するステップであって、前記第2遅延値を、前記アレイから聴取者までの前記チャネルの音波の予想伝搬距離に応じて選択する、ステップと、 各出力変換器に関して、前記少なくとも1つのサラウンド・サウンド・チャネルを表す信号を遅延させた複製を得るステップであって、各遅延複製を、当該出力変換器および当該チャネルについて計算した前記第1遅延値だけ遅延させる、ステップと、 各出力変換器に関して、前記中央チャネルを表す信号を遅延させた複製を得るステップであって、各遅延複製を前記第2遅延値だけ遅延させる、ステップと、 各出力変換器に関して、前記少なくとも1つのサラウンド・サウンド・チャネルを表す前記信号の前記遅延複製を、前記中央チャネルを表す前記信号のそれぞれの遅延複製に加算する、ステップと、 前記出力変換器のアレイを用いて、前記遅延複製を出力するステップと、 から成り、 聴取者に到達する前に、少なくとも1つのチャネルの音波を壁のような表面で跳ね返す、 方法。
- 8請求項 7 記載の方法であって、更に、 前記中央チャネルに対して、各出力変換器に関して第1遅延値を選択するステップであって、前記第1遅延値を、前記それぞれの変換器の前記アレイにおける位置に応じて選択し、前記中央チャネルを所定の方向に送出する、ステップを含み、 各出力変換器に関して前記中央チャネルを表す信号を遅延させた複製を得るステップは、更に、 前記中央チャネルを表す前記信号の各複製を、前記それぞれの出力変換器および当該中央チャネルについて計算した前記第1遅延値だけ遅延させるステップを含む、方法。
- 9請求項 7 記載の方法において、前記中央チャネルを表す前記信号の複製は、前記第2遅延値以外の値では遅延させず、前記第2遅延値が、前記信号の各複製について同一である、方法。
- 10請求項 7 ないし 9 のいずれか1項に記載の方法であって、更に、 前記少なくとも1つのサラウンド・サウンド・チャネルに対して、各出力変換器に関して第2遅延値を選択するステップであって、前記第2遅延値を、前記アレイから聴取者までの前記チャネルの音波の予想伝搬距離に応じて選択するステップを含み、 各出力変換器に関して、前記少なくとも1つのサラウンド・サウンド・チャネルを表す信号を遅延させて複製を得る前記ステップは、更に、 前記少なくとも1つのサラウンド・サウンド・チャネルを表す前記信号の各複製を、前記それぞれの出力変換器および前記少なくとも1つのサラウンド・サウンド・チャネルについて計算した前記第2遅延値だけ遅延させるステップを含む、方法。
- 11請求項 7 ないし 10 のいずれか1項に記載の方法において、前記信号を複製する前に、前記中央チャネルを表す各信号に前記第2遅延を与える、方法。
- 12請求項 7 ないし 11 のいずれか1項に記載の方法において、前記音場が、2つのサラウンド・サウンド・チャネルを備え、各サラウンド・サウンド・チャネルを異なる方向に送出する、方法。
- 13請求項 7 ないし 12 のいずれか1項に記載の方法において、前記第2遅延値の選択を、全サウンド・チャネルの対応する部分が実質的に同時に聴取者に到達するように行う、方法。
- 14音場を作り出す装置であって、 少なくとも1つのサラウンド・サウンド・チャネルと中央チャネルとを表す複数の入力信号を受ける手段と、 出力変換器のアレイと、 各出力変換器に関して、前記少なくとも1つのサラウンド・サウンド・チャネルを表す前記信号の複製と、中央チャネルを表す前記信号の複製とを得るように構成された複製手段と、 前記少なくとも1つのサラウンド・サウンド・チャネルを表す前記信号の各複製を、前記それぞれの変換器の前記アレイにおける位置に応じて選択したそれぞれの第1遅延値だけ遅延させ、前記チャネルを所定の方向に送出するように構成された第1遅延手段と、 前記中央チャネルを表す前記信号の各複製を、前記アレイから聴取者までの前記チャネルの音波の予測伝搬距離に応じて選択した第2遅延値だけ遅延させるように構成された第2遅延手段と、 各出力変換器に毎に設けた加算手段であって、前記少なくとも1つのサラウンド・サウンド・チャネルを表わす前記信号の前記遅延複製を前記中央チャネルを表わす前記信号の前記複製の各々に加算して各出力変換器への出力信号を生成する加算手段と、 を備えてい て、 前記少なくとも1つのチャネルの音波の予想伝播距離は、壁のような表面での跳ね返りを含む、前記音場を作り出す 装置。
- 15請求項 14 記載の装置において、前記第1遅延手段が、更に、前記中央チャネルを表す前記信号の各複製を、前記それぞれの変換器の前記アレイにおける位置に応じて選択したそれぞれの第1遅延値だけ遅延させ、前記中央チャネルを所定の方向に送出するように構成されている、装置。
- 16請求項 14 または 15 記載の装置において、前記第2遅延手段が、更に、前記少なくとも1つのサラウンド・サウンド・チャネルを表す前記信号の各複製を、前記アレイから聴取者までの前記チャネルの音波の予想伝搬距離に応じて選択したそれぞれの第2遅延値だけ遅延させるように構成されている、装置。
- 17請求項 14 ないし 16 のいずれか1項に記載の装置において、前記第2遅延手段が、前記複製手段によって複製する前に、前記入力信号を遅延させるように構成されている、装置。
- 18請求項 14 ないし 17 のいずれか1項に記載の装置において、前記音場が2つのサラウンド・サウンド・チャネルを備えており、前記第1遅延手段が、各サラウンド・サウンド・チャネルを異なる方向に送出させるように構成されている、装置。
- 19請求項 14 ないし 18 記載のいずれか1項に記載の装置において、前記第2遅延手段が、全サウンド・チャネルが実質的に同時に聴取者に到達するように、前記チャネルに対して前記第2遅延を選択するように構成されている、装置。
- 20請求項 14 ないし 19 のいずれか1項に記載の装置において、前記第1遅延手段および前記第2遅延手段が、同じ物理的手段である、装置。
- 21請求項 7 ないし 13 のいずれか1項に記載の方法または請求項14ないし20のいずれか1項に記載の装置において、前記出力変換器をクラス-BD PWM増幅器によって直接駆動する、方法または装置。
Independent claims21
111 paragraphs, as filed
The present invention relates to a steerable acoustic antennae, and more particularly to a digital electronic steerable acoustic antenna.
Phased array antennas are well known in the art in both electromagnetic and ultrasonic acoustic fields. These are lesser known, but in simple form, they also exist in the field of acoustic (audible) acoustics. These latter are relatively immature, and the present invention seeks to make improvements to the finest audio-acoustic arrays that can be steered to deliver output to some extent at will.
<p> WO96 / 31086 describes a system that drives an array of output transducers using unary coded signals. Each transducer can produce sound pressure pulses and cannot reproduce the entire output signal.</p>
<p> A first aspect of the invention addresses a possible problem when a single output converter array outputs a large number of channels and directs each channel in different directions. Due to the fact that each channel takes a different path to the listener, it is possible to hear the channel out of sync as it reaches the listener's position.</p><p> According to the first aspect, a method of creating a sound field consisting of a plurality of sound channels is provided by using an array of output converters. A step of selecting a first delay value for each output converter for each channel, the step of selecting the first delay value according to the position of the respective converter in the array. A step of selecting a second delay value for each channel, wherein the second delay value is selected according to the expected propagation distance of sound waves of the channel from the array to the listener. For each output converter, a step of obtaining a delayed replication of a signal representing each channel, each delayed replication being composed of a first component consisting of the first delay value and a second component consisting of the second delay value. Delay by the value that has, step and Consists of.</p><p> Further, according to the first aspect of the present invention, a device for creating a sound field is provided, and the device is With multiple inputs for each of the multiple signals representing different sound channels, Output converter array and With respect to each output converter, a duplication means configured to obtain duplication of each input signal, A first delay means configured to delay each replication of each signal by each first delay value selected according to the position of each output converter in the array. From the second delay means configured to delay each replication of each signal by a second delay value selected for each channel according to the expected propagation distance of the sound waves of the channel from the array to the listener. Become.</p><p> Therefore, there are provided methods and devices for providing two types of delay to each sound channel to mitigate the effects of different propagation distances for each channel.</p><p> A second aspect of the present invention addresses a problem that arises when applying an array of output transducers to audio-visual. Often it is necessary to give the channel various delays to produce the desired effect, which can cause the sound channel to lag significantly behind the video footage.</p><p> According to a second aspect of the present invention, in an audio-visual presentation, a method of temporally associating video and sound and using an array of output converters to reproduce sound content consisting of a plurality of channels. Is provided and the method is For each output converter, the step of delaying the replication of each signal representing the sound channel by the respective audio delay value, It consists of a step of delaying the video signal by the video delay value calculated so that the corresponding video image is displayed at a substantial time when the corresponding sound channel reaches the listener in time.</p><p> Further, according to a second aspect of the present invention, there is provided a device for temporally associating a video with a plurality of sound channels in an audio-visual presentation. Output converter array and For each output converter, replication and delay means configured to obtain a delayed replication of each signal representing the sound channel. Video delay means configured to delay a video signal by the video delay value calculated to display the corresponding video footage at a substantial time when the corresponding sound channel reaches the listener in time. When, It has.</p><p> Thus, this aspect of the invention allows video and sound channels to reach the viewer at the correct time (ie, temporally corresponding to each other).</p><p> A third aspect of the invention addresses the problem that different sound channels can differ in content and therefore have different requirements regarding the directivity to be achieved by any of the individual beams representing the sound channel.</p><p> To this end, a third aspect of the invention provides a method of creating a sound field consisting of a plurality of sound channels using an array of output transducers, the method of which is: For each channel, for each output converter, a step of obtaining a copy of the signal representing the channel and obtaining a set of duplicate signals for each channel. The step of applying the first window function to the first set of duplicate signals expressed from the first sound channel signal, It consists of a step of applying a different second window function to the second set of duplicate signals expressed from the second sound channel signal.</p><p> Further, according to a third aspect of the present invention, a device for creating a sound field composed of a plurality of sound channels is provided, and the device is provided. Output converter array and For each output converter, a duplication means that creates a duplication of signals representing each of the plurality of channels. A window means that applies the first window function to the first set of duplicated signals generated from the first sound channel signal and applies a different second window function to the second set of duplicated signals generated from the second channel signal. It has.</p><p> Therefore, in this aspect, different window functions can be applied to different sound channels, resulting in a more desirable sound field and facilitating independent adjustment of the volume of each sound channel.</p><p> A fourth aspect of the invention addresses the problem that large arrays are required to deliver low frequencies, while smaller arrays are required to deliver high frequencies with the same accuracy. Moreover, low frequencies require more power than high frequencies.</p><p> According to a fourth aspect of the present invention, there is provided a method of creating a sound field using an array of output converters. The step of dividing the input signal into at least low frequency components and high frequency components, The step of outputting the low frequency component by using the output converter extending to the first part of the array, and It consists of a step of outputting the high frequency component by using an output converter that extends to the second part of the array, which is smaller than the first part.</p><p> Further, according to a fourth aspect of the present invention, a device for creating a sound field is provided. It comprises an array of output transducers, the transducers being mounted denser in the first area of the array than the rest of the array.</p><p> Therefore, in this aspect, it is possible to output any frequency with a desired directivity using an efficient number of output converters.</p><p> A fifth aspect of the invention relates to an efficient configuration of an array capable of delivering sound in a substantially desired plane.</p><p> According to a fifth aspect of the present invention, an array of output converters arranged linearly adjacent to each other is provided, and each of the output converters has a dimension in a direction perpendicular to the straight line. It is larger than the dimension parallel to the straight line.</p><p> The above configuration is particularly useful as the sound is mainly concentrated in a plane that extends horizontally in front of the array. Concentration on a surface is achieved by the elongated state of the individual transducers, and directivity is achieved by multiple transducers in the array.</p><p> A sixth aspect of the present invention addresses the desire to use a reflective or resonant surface to deliver a narrow or wide beam at a defined position, depending on the user's wishes.</p><p> According to a sixth aspect of the present invention, there is provided a method of making a plurality of input signals representing each channel appear to be emitted from different positions in space. A step of providing an acoustic reflection or resonance surface at each of the spatial positions, A step of providing an array of output converters away from the spatial position, Using the array of output converters, the sound waves of each channel are sent out toward their respective spatial positions, the sound waves are retransmitted by the reflecting surface or the resonant surface, and the sound waves are transmitted in front of or behind the reflecting surface or the resonant surface. The focused step of the sound wave at the spatial position and The steps to be sent consist of For each converter, delayed duplication is obtained by delaying each input signal by the respective delay amount selected according to the position of the respective output converter in the array and the respective focus position. The step of transmitting the sound wave of the channel toward the focus position with respect to the channel, and For each transducer, the step of summing the respective delayed replicas of each input signal to generate an output signal, It consists of a step of deriving the output signal to each of the converters.</p><p> Further, according to a sixth aspect of the present invention, there is provided a device that makes it appear that a plurality of input signals representing each channel are emitted from different positions in space. With the acoustic reflection or resonance surface at each of the spatial positions, With an array of output converters located away from the spatial position, Using the array of output converters, the sound waves of each channel are sent out toward their respective spatial positions, the sound waves are retransmitted by the reflecting surface or the resonant surface, and the sound waves are transmitted in front of or behind the reflecting surface or the resonant surface. A control unit that focuses the sound wave at a spatial position, The control unit consists of For each converter, each input signal is delayed by the respective delay amount selected according to the position of the respective output converter in the array and the respective focus positions to obtain delayed duplication, and the sound wave of the channel is obtained. With respect to the channel, with replication and delay means configured to deliver towards the focus position. For each converter, an addition means configured to add up the respective delayed duplications of each input signal to produce an output signal. It is provided with means for deriving the output signal to each of the converters and transmitting the sound waves of the channel toward the focus position with respect to the input.</p><p> A sixth aspect of the invention allows a narrow or wide beam to be retransmitted, depending on the focused position selected behind or in front of the reflector / resonator.</p><p> A seventh aspect of the invention can make it difficult to accurately determine where the sound is transmitted or focused, allowing the operator to control where the sound is transmitted or focused (feedback). Addresses the problem that an intuitive method is required.</p><p> According to a seventh aspect of the present invention, there is provided a method of selecting the direction in which the sound is focused. A step of determining whether or not the direction is desired using a viewfinder or other screen means and aiming the video camera in the desired direction. It consists of a step of calculating multiple signal delays given to a set of duplicates of the input signal and sending the sound in the selected direction.</p><p> Further, according to a seventh aspect of the present invention, there is provided a method of determining where to send the sound, which method. A step that automatically adjusts the direction in which the camcorder is aimed according to the direction in which the sound is emitted, It consists of a step of identifying the direction in which the camera is aimed from the viewfinder or other screen means.</p><p> Further, according to a seventh aspect of the present invention, a device for setting or monitoring a sound field is provided. Output converter array and With a variable direction video camera The output converter array and the video camera are provided with means for controlling the video camera so that the video camera is pointed in the same direction as the sound from the array is transmitted.</p><p> Therefore, in the seventh aspect of the present invention, the user can intuitively and easily determine where to transmit the sound.</p><p> In general, the invention includes a plurality of spatially distributed sonic electroacoustic transducers (SETs) arranged in a two-dimensional array, each connected to the same digital signal input via an input signal divider. The input signal is modified before feeding to each SET to obtain the desired directional effect, preferably a fully digital steerable acoustic phased array antenna (digital phased array antenna, or DPAA) Applicable to systems. The various possibilities provided here, as well as the practically preferred variants, can be seen in the description below.</p><p> It is preferable to arrange the SETs in a plane or curved surface (a Surface) rather than randomly arranging them in space. However, they may be in the form of two or more adjacent subarrays-two or more closely parallel plane or curved two-dimensional stacks located in front and behind. Within a Surface, the SETs that make up the array are preferably spaced apart, ideally to completely fill the entire antenna aperture. This is impractical for an actual set with a circular cross section, but can be achieved with a set with a triangular, square, or hexagonal cross section, that is, a cross-sectional shape that generally covers the plane. If the cross section of the SET does not cover a plane, the array can be in the form of a stack or multiple arrays, i.e. three-dimensional, to make an approximation similar to aperture filling. In this case, at least one additional surface of the SET is mounted behind at least one other such surface, and the SET in the aforementioned or rear array is forward (s). Emit between gaps in the array.</p><p> The SETs are preferably similar and ideally the same. These are, of course, acoustics, or audio devices, that can evenly cover the entire audible band, perhaps from 20Hz (or less) on the lower side to 20KHz or higher (audible band) on the higher side. Is the most preferable. Alternatively, a SET having different acoustic capabilities but collectively covering the entire desired range can be used. In this way, a large number of different SETs can be physically aggregated to form a composite SET (CSET), and even if it is not possible with individual SETs, groups of different SETs are united to form an audible band. Can be covered. As yet another variant, there is sufficient variation between SETs to completely or almost completely cover the audible band as a whole array instead of assembling the SETs, each of which only partially covers the audible band. It can also be held and scattered throughout the array.</p><p> An alternative form of CSET is to incorporate several (usually two) identical transducers, each driven by the same signal. This retains many of the advantages of large DPAAs while reducing the required signal processing and drive electronics complexity. When referring to the position of the CSET hereafter, it is understood that this position is the center of gravity of the CSET as a whole, that is, the center of gravity of all the individual SETs constituting the CSET.</p><p> Within the Surface, the spacing between SETs or CSETs (the two are simply referred to as SETs), that is, the overall layout and structure of the array, and how the individual transducers are placed within it. , It is preferable that they are regular, and their dispersion over the entire surface is preferably symmetrical. Therefore, it is most preferable that the SETs are separated in a triangular, square or hexagonal grid pattern. The type and orientation of the grid can be selected to control the spacing and orientation of the side lobes.</p><p> Although not required, each SET preferably has omnidirectional input / output characteristics, at least in the hemisphere, at a total sound wave length that can be effectively radiated (or received).</p><p> Each output SET can be in the form of any sound emitting device in any convenient or desired form (eg, conventional loudspeakers), all of which are preferably the same. Can be different. Round speakers are often in the form known as pistonic acoustic radiators (the diaphragm of the converter is moved by a piston), in which case the maximum emission of the individual SET pistons-radiators. The range (eg, the effective piston diameter for a circular SET) is preferably as small as possible, ideally as small as or less than the highest frequency acoustic wavelength in the audible band (eg, in the air). Since the wavelength of the 20 KHz sound wave is about 17 mm, the maximum diameter of the circular piston converter is preferably about 17 mm, and the size is preferably smaller than this in order to ensure omnidirectionality.</p><p> It is very likely that the overall dimensions of the SET in the plane of the array or each array will be as large as or greater than the lowest frequency acoustic wavelength in the air intended to significantly affect the polar radiation pattern of the array. Is preferable. Therefore, if it is desirable to be able to emit or steer a frequency as low as 300 Hz, the array size should be at least c in the direction perpendicular to each surface that requires steering or firing.<sub>s</sub>/ 300 1.1 meters (c<sub>s</sub>Is the speed of sound).</p><p> The present invention can be applied to fully digital steered sonic / audible acoustic phased array antenna systems, where the actual transducer can be driven by analog signals, which are driven by a digital power amplifier. Is the most preferable. A typical such digital power amplifier is a PCM signal input, a clock input (or a means of obtaining a clock from an input PCM signal), an internally generated or input clock or an output clock obtained from an additional output clock input, as well as a digital. It incorporates an optional output level input, which can be either a (PCM) signal or an analog signal (in the latter case, this analog signal can also power the amplifier output). One of the characteristics of digital power amplifiers is that they can be given discrete values to their outputs prior to any optional analog output filtering, made stepwise continuous, and level-changed only at intervals that match the output clock period. Can be mentioned. The discrete output value is controlled by the optional output level input, if provided. In a digital amplifier using PWM, the average value of the output signal represents the input signal at any integral multiple of the input sample period. In other digital amplifiers, the mean value of the output signal tends towards the mean value of the input signal over a period longer than the input sample period. Preferred forms of digital power amplifiers include bipolar pulse width modulators and 1-bit binary modulators.</p><p> The use of digital power amplifiers avoids the more common requirement of providing a digital-to-analog converter (DAC) and linear power amplifier for each converter drive channel found in most so-called "digital" systems. Therefore, the power drive efficiency can be made very high. In addition, most movable coiled acoustic transducers are inductive in nature and mechanically act very effectively as low-pass filters, so elaborate electronic low-pass filtering can be combined with digital drive circuits. It may not be necessary to add it between SET. In other words, the SET can be driven directly by a digital signal.</p><p> The DPAA has one or more digital input terminals (inputs). If there are more than one input terminal, it is necessary to provide means for deriving each input signal to each SET.</p><p> This may be done by connecting each of the inputs to each of the SETs via one or more input signal distributors. Most basically, it feeds the input signal to a single distributor, which has a separate output for each of the SETs (and, as discussed below, the signal it outputs). Appropriately modify to achieve the desired purpose). Alternatively, it may have a number of similar distributors, each taking an input signal or part of it, or a separate input signal, and each supplying a separate output to each of the SETs (and in each case it). The output signal is properly modified by the distributor to achieve the desired purpose, as discussed below). In this latter case, multiple distributors each supply to all SETs, but the output from each distributor to any one of the SETs must be combined, before further modification to the resulting signal. It is convenient to do this with an adder circuit.</p><p> The input terminal preferably receives one or more digital signals (input signals) representing the sound processed by DPAA or a plurality of sounds. Of course, the original electrical signal that defines the sound to radiate may be in analog form, so the system of the invention can include one or more analog-to-digital converters (ADCs), assisting each of these. By connecting between the analog input terminal (analog input) and one of the inputs, it is possible to convert these external analog electric signals into internal digital electric signals. Each digital electrical signal has a specific (and appropriate) sample rate Fs<sub>i</sub>Have. Therefore, within DPAA, after input, the signal processed is a time-sampled quantized digital signal, representing a sound waveform or multiple waveforms reproduced by DPAA.</p><p> The DPAA of the present invention incorporates a distributor, which corrects the input signal and then supplies it to each SET to achieve the desired directional effect. The distributor is a digital device, or piece of software. software), with one input and many outputs. One of the DPAA's input signals is fed to that input. It preferably has one output per SET, or one output can be shared by many SET or CSET elements. The distributor sends an entirely different modification of the input signal to each of its outputs. Modifications can be fixed or adjustable using a control system. Modifications made by the distributor can include providing signal delay, applying amplitude control, and adjustingly digitally filtering. These modifications may be made by a single delay means (SDM), amplitude control means (ACM), and adjustable digital filter (ADF), each located inside the distributor. It should be noted that the ADF can be configured to delay the signal with proper selection of filter coefficients. Further, this delay can be frequency dependent, and if different frequencies of the input signal are delayed by different amounts, the filter will filter the sum of any number of such delay versions of this signal. It can also be generated by effects. The term "delayed" or "delayed" as used herein shall be construed to include the form of delay given by ADF and SDM. The delay can be any useful period, including zero, but in general, at least one duplicate input signal is delayed by a nonzero value.</p><p> The signal delay means (SDM) is a variable digital signal time delay element. These are not single frequency or narrow frequency band phase shift elements, but they are true time delays, so DPAAs operate over a wide frequency band (eg, audible band). It would be nice to have a means of adjusting the delay between a given input terminal and each SET, and it would be useful to have a delay element that could be adjusted separately for each input / SET combination.</p><p> The minimum possible delay for a given digital signal is the sample period T for that signal.<sub>s</sub>Preferably less than or equal to, and the maximum delay possible for a given digital signal is the largest lateral range of sound in the transducer array D.<sub>max</sub>Time required to cross across T<sub>c</sub>It is preferable to select so as to be the same as or higher than. Where T<sub>c</sub>= D<sub>max</sub>/ c<sub>s</sub>And c<sub>s</sub>Is the speed of sound in the air. Most preferably, the minimum incremental change in delay possible for a given digital signal is T.<sub>s</sub>, Should be less than or equal to the signal sample period. Otherwise, signal interpolation will be required.</p><p> The amplitude control means (ACM) is conveniently implemented as a digital amplitude control means for the purpose of gross beam shape modification. It can increase or decrease the magnitude of the output signal if it is equipped with an amplifier or an alternator. As with SDM, it is preferable to have an adjustable ACM for each input / SET combination. The amplitude control means is preferably configured to counteract the fact that DPAAs are of finite size by applying different amplitude controls to each signal output from the distributor and by using a window function. This is conveniently done by normalizing the magnitude of each output signal according to a default curve, such as a Gaussian curve or a raised cosine curve. Thus, overall, the output signals destined for the SET near the center of the array are not significantly affected, but the SETs near the perimeter of the array are attenuated according to their proximity to the edges of the array. Will be done.</p><p> In another way to modify the signal, the digital filter (ADF) used is one in which its group delay and magnitude response is specified as a function of frequency (not just a simple time delay or level change). It changes with. These filters may be implemented with simple delay elements to reduce the required computation. This technique allows the control of the DPAA emission pattern to be performed as a function of frequency, allowing the control of the DPAA emission pattern to be adjusted separately in different frequency bands (which is the wavelength of the DPAA emission area). The size, and thus its directivity, is useful in other situations as it is a function of strong frequencies). As an example, for a DPAA with a range of 2m, for example, its low frequency cutoff (due to directivity) is around the 150Hz region, and the human ear can determine the directivity of such low frequency sounds. Because of the difficulty, it may be more useful to not apply the "beam steering" delay or amplitude weighting at such low frequencies and instead find the optimum output level. In addition, it is possible to compensate for some variation in the radiation pattern of each SET by using a filter.</p><p> The SDM delay, ACM gain and ADF coefficients can be fixed, changed in response to user input, or changed under automatic control. Preferably, any change required during use of the channel is made in many small steps so that no discontinuity is heard. These step sizes can be selected to specify predetermined "roll-off" and "attack" rates that describe how quickly the parameters can change.</p><p> If more than one input is supplied, that is, if there are I inputs numbered from 1 to I, and if there are N SETs numbered from 1 to N, then individually Separate adjustable delay, amplitude control and / or filter means D<sub>in</sub>(Here, I = 1 to I and n = 1 to N between each of the I inputs and each of the N SETs) are preferably provided for each combination. Thus, for each SET, there are I delayed or filtered digital signals, one from each of the inputs via a separate distributor, which are combined before being applied to the SET. Overall, there are N separate SDMs, ACMs and / or ADFs in each distributor, and one distributor for each SET. As noted earlier, it is convenient to use digital algebraic addition of I separate delay signals to make this digital signal combination. That is, the signal to each SET is a linear combination of separately modified signals from each of the I inputs. The need to perform digital addition of signals originating from more than one input means that it is common to perform digital addition on two or more digital signals at different clock rates and / or phases. Means that it may be necessary to use a digital sampling rate converter (DSRC) to synchronize these external signals.</p><p> The DPAA system communicates with the DPAA electronics (ideally from somewhere within the DPAA's listening area) over a distance (by wire, wireless or infrared, or some other wireless technology) and is the DPAA's mainstay. It may be used together with a remote control slave unit (slave unit) that can be manually controlled by all functions. It is most useful for such a system to have the following functions. 1) The choice of which input (multiple inputs) to connect to which distributor can also be called a "channel". 2) Control of focusing position and / or beam shape of each channel 3) Control of individual volume level settings for each channel, and 4) Initial parameter setting using a handset with a built-in microphone (see below).</p><p> Also, Means for interconnecting two or more such DPAAs and coordinating their radiation patterns, their focusing, and their optimization procedures, A means of storing and recalling a set of delays (for DDG) and filter coefficients (for ADF), There may be.</p><p> The present invention will be further described as merely a non-limiting example with reference to the attached configuration diagram.</p>
The descriptions and drawings presented below necessarily describe the invention with block diagrams, where each block represents a hardware component or signal processing step. In principle, the invention can be achieved by configuring separate physical components to perform each step and interconnecting them as shown. Some of the steps can be performed by using dedicated or programmable integrated circuits and, if possible, incorporating several steps into one circuit. It will be appreciated that in practice it is likely most convenient to perform some of the signal processing steps in software using a digital signal processor (DSP) or general purpose microprocessor. Consecutive steps can then be performed by separate software routines that share separate processors or microprocessors, or they can be incorporated into a single routine for increased efficiency.
The drawings only outline the audio signal path, and clock and control connections are omitted for clarity except when necessary to convey ideas. Furthermore, only a small number of SETs, channels, and circuits associated with them are shown, because if a large number of elements are realistically included, the drawing becomes complicated and difficult to interpret.
Before describing each aspect of the invention, it is useful to describe embodiments of the device suitable for use in any of the respective aspects.
The block diagram in Figure 1 shows a simple DPAA. The input signal (101) is supplied to the distributor (102), and the outputs of many of them (six in the figure) are each connected to the output SET (104) via an optional amplifier (103). The output SETs are arranged to physically form a two-dimensional array (105). The distributor modifies the signal sent to each SET to produce the desired radiation pattern. As illustrated below, additional processing steps may be provided before or after the distributor.
FIG. 2 shows a DPAA with two input signals (501, 502) and three distributors (503-505). Distributor 503 processes signal 501, while both 504 and 505 process input signal 502. The output from each distributor addressed to each SET is added by the adder (506) and reaches the SET 104 via the amplifier 103.
Figure 3 shows the components of the distributor. It has a single input signal (101) coming from the input circuit and a large number of outputs (802), one for each SET or SET group. The paths from each of the inputs to the outputs include SDM (803) and / or ADF (804) and / or ACM (805). If the modifications made in each signal path are similar, the distributor can be implemented more efficiently by including a common SDM, ADF and / or ACM (806-808) before splitting the signal. .. Each parameter of each distributor part can be changed by the user or under automatic control. The control connections required for this are not shown.
FIG. 4 shows a possible power amplifier configuration. In one option, the input digital signal (1001), perhaps from a distributor or adder, passes through a DAC (1002), and a linear power amplifier (1003) with an optional gain / volume control input (1004). The output is sent to SET or SET group (1005). In a suitable configuration when exemplifying two SET feeds, the input (1006) is sent directly to a digital amplifier (1007) with an optional global volume control input (1008). Be done. The comprehensive volume control input is convenient because it can also function as a power source for the output drive circuit. The output of a digital amplifier that takes discrete values also has the option of passing through an analog low-pass filter (1009) before reaching SET (1005).
Figure 5 shows the interconnection of three DPAAs (1401). In this case, the input (1402), input circuit (1403) and control system (1404) are shared by all three PDAAs. The input circuit and control system can be housed separately or incorporated into one of the DPAAs and the other can act as a slave. Alternatively, it is possible to make the three DPAAs the same and make only the redundant circuit in the slave DPAA inactive. This setting allows for increased power and improves directivity at low frequencies when juxtaposed arrays.
The devices of FIGS. 6 and 7A to 7D have the general structure shown in FIG. FIG. 6 shows the preferred distributor (102) in more detail.
As can be seen from FIG. 6, the input signal (101) is led out to the replicator (1504) by the input terminal (1514). The replicator (1504) has a function of copying an input signal a predetermined number of times and supplying the same signal to the predetermined number of output terminals (1518). Each copy of the input signal is then fed to means for modifying the copy (1506). In general, replication-correcting means (1506) include signal delay means (1508), amplitude control means (1510), and adjustable digital filter means (152). However, it should be noted that the amplitude control means (1510) is purely optional. In addition, one or the other of the signal delay means (1508) and the adjustable digital filter (1512) may also be removed. The most basic function of the means of modifying replication (1506) is, in a sense, to supply all the different replications with different amounts of delay. It is the choice of delay that determines the sound field obtained when the output transducer (104) outputs the input signal (101) with various delays. Delayed and preferably otherwise modified replicas are output from the distributor (102) via the output terminal (1516).
As already mentioned, the choice of delay obtained by each signal delay means (1508) and / or each adjustable digital filter (1512) has a significant effect on the mode of the resulting sound field. In general, there are four particularly effective sound fields that can be combined linearly.
<u style="single">First sound field</u> Figure 7A shows the first sound field. An array (105) consisting of various output converters (104) is shown in plan view. Another output converter column may be placed above or below the illustrated column.
The delay given to each replica by the various signal delay means (508) is set to the same value, for example 0 (for planar arrays as shown) or a value that is a function of the surface shape (for curved surfaces). Will be done. This produces a roughly parallel "beam" of sound that represents the input signal (101). It has a crest F parallel to the array (105). Radiation in the direction of the beam (perpendicular to the crest) is much stronger than in other directions, but generally there are also "side lobes". The array (105) is assumed to have a physical range of one or several wavelengths at the sound frequency of interest. This fact means that the side lobes can be totally damped or moved by adjusting the ACM or ADF, if necessary.
The mode of operation can generally be thought of as one that simulates a conventional loudspeaker with a very large array (105). The individual transducers (104) of the array (105) all operate in phase, producing a symmetric beam whose principal direction is perpendicular to the plane of the array. The resulting sound field is very similar to that obtained with a single large loudspeaker with a diameter of D.
<u style="single">2nd sound field</u> The first sound field can be considered as a concrete example of the more general second sound field. Here, the delay given to each replica by the signal delay means (1508) or the adjustable digital filter (1512) is systematically delayed between the transducers (104) in a selected direction across the surface of the array. It is changing to increase. This is shown in Figure 7B. The delays given to the various signals before they are derived to their respective output converters (104) can be visualized in FIG. 7B by the dotted lines extending behind the converters. The longer the dotted line, the longer the delay time. In general, the relationship between the dotted line and the actual delay time is d<sub>n</sub>= t<sub>n</sub>* c, where d represents the length of the dotted line, t represents the amount of delay given to each signal, and c represents the speed of sound in the air.
As can be seen from FIG. 7B, the delay given to the output converter increases linearly from left to right in FIG. 7B. Therefore, the signal derived to the transducer (104a) has virtually no delay and is therefore the first signal to exit the array. The signal derived to the transducer (104b) is given a small delay, so this signal exits the array second. Since the delay given to converters (104c, 104d, 104e, etc.) increases continuously, there is a fixed delay between the outputs of adjacent transducers.
This series of delays produces a roughly parallel sounding "beam" similar to that produced in the first sound field, but here it depends on the amount of systematic delay increase used. The difference is that the beam makes an angle by the amount. Very small delay (t<sub>n</sub><< T<sub>c</sub>At, n), the beam direction is approximately orthogonal to the array (105), and increasing the delay (max t).<sub>n</sub>) ~ T<sub>c</sub>, Can be guided so as to be approximately tangential to the surface.
As described above, the sound wave is generated by selecting the delay so that the same temporal part (the part of the sound wave representing the same information) of the sound wave from each transducer together forms a wave front F propagating in a specific direction. It can be sent without focusing.
Side lobes (finite array) in the radiation pattern by reducing the amplitude of the signal given by the distributor to the SET located near the edge of the array (relative to the amplitude given to the SET near the center of the array). -The level (depending on size) can be reduced. For example, a Gaussian or raised cosine curve can be used to determine the amplitude of the signal from each SET. A trade-off is achieved between adjusting for the effects of a finite array size and reducing power by reducing the amplitude at the outer SET.
<u style="single">Third sound field</u> When selecting the signal delay given by the signal delay means (1508) and / or the adaptive digital filter (1512), a delay was added to the sound propagation time from the SET (104) to the selected point in the space in front of the DPAA. If the sum is the same for all SETs, that is, if the sound waves from each of the output converters arrive as in-phase sound at the selected point, the DPAA will focus the sound at that point P. Can be made to. This is shown in Figure 7C.
As can be seen from FIG. 7C, the delay given in each of the output converters (104a to 104h) also increases, but in this case it is not linear. This creates a curved crest F, which converges at the focus point, so that at this focus and its surroundings (a region of dimensions approximately equal to each wavelength of the spectral component of the sound). The intensity of the sound is significantly higher than at other points in the vicinity.
The calculations required to obtain the focused sound waves can be generalized as follows.<maths num="1"><img file="JP4445705B2_D0001.tif" /></maths><maths num="2"><img file="JP4445705B2_D0002.tif" /></maths><maths num="3"><img file="JP4445705B2_D0003.tif" /></maths><maths num="4"><img file="JP4445705B2_D0004.tif" /></maths>Where k is a constant offset to ensure that all delays are positive and therefore feasible.
The focal position can also be extensively changed almost anywhere in front of the DPAA by properly selecting a set of delays as described above.
<u style="single">4th sound field</u> FIG. 7D shows the fourth sound field when yet another theory is used to determine the delay given to the signal derived to each output transducer. In this embodiment, Huygens' wavelet theory is called to simulate a sound field with an apparent origin O. To do this, set the signal delay obtained by the signal delay means (1508) or adaptive digital filter (1512) to be equal to the sound propagation time from a point in the space behind the array to each output transducer. To do. These delays are shown by the dotted lines in Figure 7D.
From FIG. 7D, it can be seen that the output converter located closest to the simulated origin position outputs the signal before the converter located away from the origin position. The interference pattern formed by the waves emitted from each of the transducers creates a sound field, which appears to listeners in the near field in front of the array to originate from the simulated origin.
Figure 7D shows a hemispherical crest. The wave front F obtained by summing these has the same curvature and moving direction as the wave head has when it is emitted at the pseudo origin. In this way, a true sound field is obtained. The formula for calculating the delay here is:<maths num="5"><img file="JP4445705B2_D0005.tif" /></maths>Where t<sub>n</sub>Is defined in the third embodiment, and j is an arbitrary offset.
Therefore, in the general method used here, an replicator (1504) is used to obtain N duplicate signals. One for each of the N output transducers. Each of these replicas is then delayed (perhaps with a filter) by the amount of delay selected depending on both the location of each output transducer in the array and the effect obtained. Next, the delayed signal is derived to each output converter to create a corresponding sound field.
Preferably, the distributor (102) provides separate replication and delay means to replicate the signal so that each replication is delayed. However, other configurations are also included in the present invention, for example, an input buffer having N taps may be used, and the delay amount may be determined by the position of the taps.
Since the system described above is linear, any of the four effects described above can be combined by simply adding the required delay signals to each other for a particular output converter. Similarly, the linear nature of the system means that several inputs can be focused or oriented as described above, each separately and in a different way, and are controllable and latent. This means that a widely separable region can be obtained, in which different sound fields (representing signals at different inputs) can be established away from the DPAA itself. For example, the first signal can be generated so that it feels like it is coming from some distance behind the DPAA, and the second signal can be focused at some distance in front of the DPAA.
<u style="single">The first aspect of the present invention</u> A first aspect of the invention relates to the use of DPAA in a multichannel system. As described above, the same array can be used to deliver different channels in different directions to obtain special effects. FIG. 8 illustrates this roughly in plan view, using an array (3801) to deliver the first beam of sound (B1) forward substantially linearly towards the listener (X). are doing. It can be focused or unfocused, as shown in Figure 7A or Figure 7B. Since the second beam (B2) is emitted at a slight angle, the beam bypasses the listener (X) and reflects off the wall (3802) multiple times, again eventually to the listener. To reach. Since the third beam (B3) is transmitted at a larger angle, it bounces once on the side wall and reaches the listener. One of the typical uses of such a system is a home cinema system. In that case, beam B1 represents the central sound channel, beam B2 represents the right surround (right rear speaker in traditional systems) sound channel, and beam B3 represents the left sound channel. In addition, right and left surround channel beams are possible, but are excluded from Figure 8 for clarity. Obviously, the distance propagated before reaching the user varies from beam to beam. For example, the central beam may propagate 4.8 meters, the left and right channels may propagate 7.8 meters, and the surround channels may propagate 12.4 meters. With this in mind, it is possible to allow each channel to reach the user at substantially the same time by giving the channel with the shortest propagation distance an additional delay.
The device that accomplishes this is shown in FIG. Input three channels (3901, 3902, 3903) to their respective delay means (3904). The delay means (3904) delays the time of each channel by an amount determined by the delay control unit (3909). The delayed channel is then passed to the distributor (3905), adder (3906), amplifier (3907) and output converter (3908). The distributor (3905) creates a copy, delays the copy, and sends the channel out in different directions, as shown in FIG. The delay control unit (3909) selects the delay based on the expected distance that the sound waves of the channel propagate before reaching the user. Using the example above, the surround channel propagates the longest distance and therefore does not delay at all. The left channel is delayed by 13.5ms so that it arrives at the same time as the surround channel, and the central channel is delayed by 22.4ms so that it arrives at the same time as the surround channel and the left channel. This ensures that all channels reach the listener at the same time. When changing the direction of the channel, the delay control unit (3909) can take this into consideration and adjust the delay accordingly. In FIG. 9, the delay means (3904) is shown in front of the distributor. However, it is also effective to incorporate these into the distributor so that the delay control unit (3909) inputs the signal to each distributor and gives this delay to all the duplicated signals output by the distributor. Yet another practical alternative is to use a single delay controller (3909), select the delay obtained for each channel replication, and eliminate the need for a separate delay element (3904) for each distributor. Delayed data can be sent to.
<u style="single">Second aspect of the present invention</u> In the first aspect described above, the delay in the sound reaching the user can be significantly increased and becomes more noticeable as the magnitude is increased. In audio-visual applications, this can cause the video to precede the sound and have an unpleasant effect. This problem can be solved by using the device shown in FIG. The corresponding audio and video signals are sourced from sources such as the DVD player (4001). These signals are read out at the same time and have a temporal correspondence. A channel splitter (4004) is used to obtain each audio channel from the audio signal and apply each channel to the apparatus shown in FIG. An audio delay controller (3909) is connected to the video delay means (4005) to delay the video signal by an appropriate amount so that the sound and video can reach the user at the same time. Next, the output from the video delay means is output to the screen means (4006). The video delay given is generally calculated relative to the maximum distance the sound beam propagates, that is, the surround channel in FIG. The video delay in this case is set equal to the propagation time of beam B2 not delayed by the audio delay means (3904). It is usually desirable for the video signal to be delayed by an integer number of frames. This means that the video delay value is only approximately equal to the calculated value. Even in surround channels, any processing they receive (eg, filtering) can cause some delay. Therefore, another component may be added to the video delay value to take into account this processing delay. In addition, it is often easier to delay the video signal until the sound that reaches the listener directly on the path (eg, beam B1 in FIG. 8) is emitted from the speaker. The resulting error is generally small, and listeners are accustomed to it from current AV systems. Claims 11 and 16 are "substantially at said time" (at).
As an improvement, the video delay means can be connected to each distributor (3905) as well (see the dotted line in Figure 10), thus giving due consideration to any delay given due to the directivity of the beam. Can be done. As another improvement, a video processing circuit can be used to provide an on-screen display of the sound system's user interface. In a more general software embodiment, each component of the audio delay is calculated by the microprocessor as part of the program and the complete delay value is calculated for each replication. These values are then used to calculate the appropriate video delay.
<u style="single">Third aspect of the present invention</u> When using a large number of channels, it may be effective to apply different window functions to each channel. The window function reduces the effects of "side lobes" at the expense of power. The type of window function to be used is selected according to the required amount of the beam to be obtained. Therefore, when the directivity of the beam is important, it is advisable to use the window function as shown in FIG. 11A. If the directivity requirements are low, a looser function as shown in Figure 11D can be used.
A device for accomplishing this is shown in FIG. This device is substantially identical to that shown in FIG. 9, except that the additional delay means (3904) has been removed. Such additional delaying means, however, can be combined with this aspect of the invention. In Figure 12, an additional component (4101) is placed after the distributor. This component gives a window function. This component can be effectively combined with a distributor, but is shown separately for clarity. Window means (4101) applies window functions to a set of duplicates for a channel. Therefore, the system can be configured to select different window functions for each channel.
The system has yet another advantage. Channels with high bass content generally need to have high levels, but directivity is less important. Therefore, for such channels, the window function can be modified to meet these needs. An example is shown in FIGS. 11A to 11D. Figure 11A shows a typical window function. The transducer near the outside of the array (4102) has a lower output level than the transmitter in the center to reduce side lobes and improve directivity. Increasing the volume increases the total output level, saturates some of the transducers in the center of the array (see Figure 11B), and causes full scale (FSD). There is a risk of leading to deflection). To avoid this, the shape of the window function can be changed instead of simply amplifying the output of each transducer. This is shown in FIGS. 11C and 11D. As the volume is increased, the outer transducers play a more important role in contributing to the overall sound. This also increases the side lobes, but also increases the power output, resulting in a louder sound with no clipping at all.
The techniques described above are most important for the higher frequency components. Therefore, it may be advantageous to combine this aspect with a fourth aspect (shown below). At lower frequencies, directivity is less achieved and less important, so a flat (boxcar) window function can be used to obtain maximum power output. In addition, it is not essential to modify the window function in consideration of the volume increase as shown in FIG. 11D, and in the saturation shown in FIG. 11B, the window is similarly lowered to 0 to avoid the discontinuity at the edge. And since the level discontinuity is more detrimental than the tilt discontinuity, the sound quality can actually not be noticeably degraded.
<u style="single">Fourth aspect of the present invention</u> The directivity achievable by the array is a function of the frequency of the signal being sent and the size of the array. Sending a low frequency signal requires a larger array than sending a high frequency signal with the same resolution. Moreover, low frequencies generally require more power than high frequencies. Therefore, it is advantageous to divide the input signal into two or more frequency bands and treat these frequency bands separately with respect to the directivity achieved using the DPAA device.
FIG. 13 shows a typical device that selectively emits different frequency bands. The input signal 101 is connected to a signal splitter / combiner (2903) and is therefore connected to a low-pass filter (2901) and a high-pass filter (2902) in a parallel channel. The low pass filter (2901) is connected to the distributor (2904), the distributor (2904) is connected to all adders (2905), while the adder (2905) is of PDAA (105). It is connected to N converters (104).
The high-pass filter (2902) is connected to the same device (102) as the device (102) in FIG. 1 (and with approximately N variable amplitude and variable time delay elements built-in). Meanwhile, the device (102) is connected to another port on the adder (2905).
The system can be used to overcome the effects of these low frequency far field cancellations due to the small size of the array compared to the wavelengths at low frequencies. Therefore, the system can process different frequencies individually with respect to the shaping of the sound field. The low frequencies pass between the source / detector and the transducers (2904), all with the same time delay (0 on the surface) and amplitude, while the high frequencies pass through each of the N transducers. In contrast, the amplitude is controlled independently with an appropriate time delay. This allows for anti-beaming, or high frequency loss, without causing global far-field nulling of the low frequencies.
It should be noted that the method according to the fourth aspect of the present invention can be carried out using an adjustable digital filter (512). Such filters allow different delays to be harmonized to different frequencies by simply selecting the appropriate value for the filter factor. In this case, it is not necessary to divide the frequency bands separately and give different delays to the replications obtained from each frequency band. Appropriate effects can be obtained by simply filtering various duplicates of a single input signal.
FIG. 14 shows another embodiment of this embodiment. Here, the output converters of the array are used as different sets to transmit different frequency bands of the input signal (101). As shown in FIG. 13, the input signal (101) is divided into a high frequency band by a high pass filter (3402) and a low frequency band by a low pass filter (3405). The low frequency signal is derived to the first transducer set (3404) and the high frequency band is derived to the second transducer set (3405). The physical range of the array occupied by the first transducer set (3404) is larger than that of the high frequency transducer (3405). Generally, the range occupied by the transducer set (ie, the magnitude of the characteristic dimensions) is roughly proportional to the shortest wavelength to be transmitted. This gives roughly equal directivity to both (or all, if more than two) frequency bands.
FIG. 15 shows yet another embodiment of this embodiment. Here, a part of the output converter is shared between bands. Again, the signal is divided into low and high frequency components by a low pass filter (3501) and a high pass filter (3502). The low frequency divider (3503) derives a properly delayed replica of the low frequency component of the input signal to the first output transducer set (3505). In this example, this first set consists of all the transducers in the array. The high frequency divider derives the high frequency component of the input signal into the second output converter set (3506). These transducers are a subset of the entire array and may be the same as those used to output the low frequency components, as shown in the figure. In this case, an adder (3504) is required to add the low and high frequency signals before output. Therefore, in this embodiment, more converters are used to output the low frequency components and thus more power is needed at low frequencies. In order to further improve the power output at low frequencies, the outer transducer (which outputs only low frequencies) can be made larger and more powerful.
This method has the advantage that the resulting directivity is the same across all frequencies, minimizing the number of transducers used at high frequencies and, as a result, reducing complexity and cost. This is especially true when the low frequency dedicated converter is located outside the array and the high frequency converter is near the center, using the settings shown in FIG. Further, this has an advantage that an inexpensive range-limited converter can be used instead of the full-range converter.
FIG. 16 schematically shows a front view of the transducer array, with each symbol representing the transducer (note that the symbols are not intended to be related to the shape of the transducer used at all. deep). When using the method of FIG. 14, the square symbol represents the transducer used to output the low frequency components. The circular symbol represents a converter that outputs an intermediate range component, and the triangular symbol represents a converter that outputs a high frequency component.
When using the method of FIG. 15, the triangular symbol represents a transducer that outputs components in all three frequency ranges. The circular symbol represents a converter that outputs only intermediate range and low frequency signals, and the square symbol represents a converter that outputs only low frequencies.
This aspect of the invention is perfectly compatible with the third aspect described above. This is because the window function can be used and the calculation is done after the distributors (3403, 3503, 3507). When using a dedicated transducer (as in Figure 14), the "holes" created in the low frequency window function due to the presence of the central array of high frequency transducers are usually not detrimental to performance. Especially when the pores are small enough for the shortest wavelength reproduced by the low frequency channel, it is not harmful.
As is clear from FIG. 16, the high frequencies use fewer transducers than the low frequencies, and the spacing between adjacent transducers is constant. However, the maximum allowable converter spacing is a function of wavelength, so converters need to be more densely packed (eg, at λ / 2 spacing) to avoid side lobes at high frequencies. This is costly for converters and drive electronics because, on the one hand, a sufficiently large area is secured to deliver low frequencies, and on the other hand, the transducers are placed in close proximity to deliver high frequencies. To solve this problem, the array shown in FIG. 17 is provided. In this array, the output converters located near the center are denser than average. Therefore, higher frequencies can be output using more dense transducers, without increasing the range of the array and thus increasing the directivity of the beam. The converters located in the large low frequency region have low density, while the central high frequency region has increased density, optimizing cost and performance at all frequencies. In FIG. 17, the square merely indicates the presence of the transducer, not the shape or type of signal output as in FIG.
<u style="single">Fifth aspect of the present invention</u> FIG. 18 shows a transducer whose length L is longer than its width W. This transducer can be effectively used in an array of similar transducers as shown in FIG. Here, the transducers 3701 are arranged linearly adjacent to each other, and the straight lines extend in a direction perpendicular to the longer side of each transducer. The sound field obtained by this arrangement can be effectively transmitted to the horizontal plane, and since the shape of each transducer is elongated, most of its energy is contained in the horizontal plane. Highly efficient operation is obtained because there is almost no sound energy transmitted to the other side. Thus, the fifth aspect provides a one-dimensional array composed of elongated transducers, giving strong directivity in one direction (due to the elongated shape) and other directions (due to the nature of the array). Gives controllable directivity. The aspect ratio of each transducer is preferably at least 2: 2, even more preferably 3: 1 and even more preferably 5: 1. Due to the elongated shape of each transducer, the sound effect is concentrated on one surface, while the linear transducer array provides high directivity within that plane. This array can be used as an array in any of the other aspects of the invention.
<u style="single">A sixth aspect of the present invention</u> A sixth aspect of the invention relates to the use of a PDAA system to produce surround sound or stereo effects using only a single sound emitting device similar to the device described above. In particular, a sixth aspect of the present invention relates to a configuration in which different sound channels are transmitted in different directions so that sound waves collide with a reflective or resonant surface and are thereby transmitted again.
A sixth aspect of the present invention allows the observer to easily perceive a separate sound field when the DPAA is operated outdoors (or elsewhere with substantially no acoustic conditions). To do so, we address the problem of having to approach the area where the sound is focused . Otherwise, it is difficult for the observer to locate the separate sound fields created.
Placing an acoustic reflective surface, or, as an alternative, an acoustic resonator that re-radiates the absorbed incident sound energy, in the path of the sound beam will re-radiate the sound, thus effectively moving away from the DPAA. It becomes a sound source and is located in the area determined by the focusing (if any) used. When using a planar reflector, most of the reflected sound is delivered in a particular direction. If a diffuse reflector is present, the sound is re-radiated in almost all directions, on the same side of the reflector where the sound comes from the DPAA, away from the reflector. Therefore, as described above, DPAA sends a number of different sound signals representing different input signals to different regions, and such reflectors or resonators are placed in each region to produce sound from each region. Can be retransmitted to form a true multiple separated-source sound radiator system using a single DPAA of the design described herein.
FIG. 20 shows the use of a single DPAA and multiple reflective or resonant surfaces (2102) to provide the listener (2103) with multiple sound sources. Surround sound effects are audible throughout the listening area, as it is not based on psychoacoustic cues.
As described above with reference to FIG. 7A or FIG. 7B, the sound beam may or may not be in focus. The focus position can be selected either in front of, behind, or behind each reflector / resonator to obtain the desired effect. FIG. 21 schematically illustrates the effect obtained when the sound beam is focused in front of and behind the reflector, respectively. The DPAA (3301) can operate to deliver sound towards reflectors (3302 and 3303) installed in the room (3304).
When the sound beam is focused at point F1 (see Figure 21) in front of the reflector (3302), the beam narrows at the focus point and then widens. The beam continues to spread after reflections from the reflector, and the listener at point P1 hears the sound. Due to the reflection, the user perceives this sound as emanating from the imaginary focus F1'. Thus, the listener at P1 perceives the sound as coming from outside the room (3304). In addition, the resulting beam is so wide that most listeners in the lower half of the room (3304) will hear this sound.
When the sound beam is focused at point F2 (see Figure 21) behind the reflector (3303), the beam reflects back toward the focus point before it is maximally narrowed. After the reflection, the beam spreads and the listener at point P2 can hear this sound. Due to the reflection, the user perceives this sound as coming from a reflected focal point F2'in front of the reflector. In this way, the listener at P1 perceives the sound as coming from a close position. Moreover, the resulting beam is so narrow that it is possible to deliver sound to only a small percentage of the listeners in the room. Therefore, for the reasons mentioned above, it can be said that focusing the beam at a position other than the reflector / resonator is effective.
As described above for a large number of isolated beams, i.e., when operating the DPAA to deliver sound signals representing different input signals to different isolated regions, a hard interface and / or sound reflection. In non-anechoic conditions with many well-characterized interfaces (normal room environment), the observer recognizes his normal sound orientation, especially if these areas are delivered to one or more reflective interfaces. Separate sound fields can be easily perceived using only abilities, and at the same time, separate in each of these spaces so that the reflected sound (from the boundary) reaches the observer from these areas. Each of these can be identified in the focal regions (if any).
In such cases, the observer perceives the actual separated sound field, but emphasizes that DPAA does not rely on introducing artificial psychoacoustic elements into the sound signal. Is important. Therefore, the observer's position is relatively insignificant in identifying the true sound, as long as it is far enough away from the near field radiation of the DPAA. In this way, it is possible to obtain a multi-channel "surround sound" using only one physical loudspeaker (DPAA) and taking advantage of the natural boundaries found in most real-world environments.
To achieve a similar effect in an environment lacking a suitable natural reflection boundary, a similarly separated multi-source sound field can be obtained by properly arranging artificial reflection or resonant surfaces. In this case, it is desirable that the sound source appears to emit a beam from these surfaces. For example, in a large concert hall or outdoor environment, a translucent plastic or glass panel can be used as a sound reflector with little visual effect. If it is desirable for the sound to be widely dispersed from these areas, a sound scattering reflector or broadband resonator can be introduced instead (this makes it more difficult to make it translucent, but not impossible. ).
Diffuse reflection can be obtained over a wide angle using a spherical reflector. In order to further enhance the diffuse reflection effect, the surface should have a roughness of about the wavelength of the sound frequency that should be diffused. A major advantage of this aspect of the invention is that all of the above can be achieved with a single DPAA device, and for each converter the output signal can be constructed by summing the delayed duplication of the input signal. This eliminates the need for many wires and devices traditionally associated with surround sound systems.
<u style="single">Seventh aspect of the present invention</u> A seventh aspect of the invention states that it is not always easy for a DPAA user to identify where the sound of a particular channel is being transmitted or focused at any particular time point. Address the problem. Conversely, the user may want to send or focus the sound at a specific location in space, but may require complex calculations regarding the correct delay to be given, etc. This problem is mitigated by providing video camera means that can be aimed in a particular direction. Means connected to the video camera can then be used to calculate the aiming direction of the camera and adjust the delay accordingly. Advantageously, the camera is under the direct control of the operator (eg, on a tripod or with a joystick), and the PDAA control unit is wherever the operator attempts to aim the camera. , It is configured to send out a sound channel. This makes it very easy to install a system that does not rely on creating mathematical models of rooms or performing other complex calculations.
It is convenient to provide a means to detect where in the room the camera is in focus. The sound beam can then be focused on the same spot. This makes the system very easy to set up. This is because markers can be placed in the room where you want the sound to be focused, and then the operator can focus the camera lens on these markers while watching the television monitor. Is. In addition, the system can automatically configure the software to calculate the correct delay for focusing the sound on that spot. Alternatively, you can identify a reference point in the room and select the focus of the sound. For example, a simple model of a room can be pre-programmed so that the operator can select an object within the field of view of the camera and determine the focal length. Coordinates from the camera (pan, tilt, distance) or room (x, y, z) to the speaker (rotation, elevation, distance), both when using the focal length of the camera and when using the model of the room. It is convenient to use conversion. In this case, the two coordinate shapes have different origins.
In the reverse mode of operation, the PDAA's electronics can also automatically steer the camera so that the beam is aimed in the direction it is currently steered, no matter what. Automatically focuses on the point where sound focusing occurs. This provides the operator with a large amount of useful configuration feedback information.
In addition, it is preferable to provide a means for selecting which channel setting is controlled by the camera position, and it is preferable to control all of them from the slave unit.
FIG. 22 is a side view showing a case where a video camera (3602) located on the DPAA (3601) is used to aim at the same point where the sound is in focus. The camera can be steered using a servo motor (3603). Alternatively, the camera can be mounted on a separate tripod, held by hand, or part of an existing CCTV system.
In CCTV applications, multiple cameras are used to correspond to one range, but a single array can be used to deliver sound to any position within this range that one of the cameras aims at. it can. Therefore, the operator selects a camera that is aimed at that point and speaks into the microphone to deliver sound (such as a voice command or command) to a specific point in the range / room. be able to.
<u style="single">More preferred features</u> Means may be provided to adjust the emission pattern and focus point of the signal for each input in response to the value of the program digital signal at these inputs. Using such a technique, if there is a large sound to be played only from that input, the stereo signal and surround sound effect can be exaggerated by temporarily moving the focus point of these signals outward. .. In this way, steering can be performed according to the actual input signal itself.
In general, when moving the focus point, it is necessary to change the delay given to each copy, which involves copying or skipping the sample as appropriate. Preferably, this is done in stages to avoid audible clicks. Click sounds can occur, for example, when skipping a large number of samples at once.
Practical application areas of the technique of the present invention include: Home entertainment can project many real sources of sound to different locations in the listening room, creating clutter, complexity, and wiring problems when wiring multiple separate loudspeakers. It enables playback of multi-channel surround sound.
In PA (public address) and concert sound systems, the emission pattern of DPAA can be freely changed in three dimensions, and a large number of simultaneous beams can be obtained. Very fast settings, like the DPAA's physical orientation, are less important and do not need to be adjusted repeatedly. A wide variety of radiation patterns can be obtained with a smaller assortment of loudspeakers, such as one type of speaker (DPAA). This usually requires dedicated speakers, each with a suitable horn. Only by adjusting the filter and delay factor can the sound energy reaching the reflective surface be reduced, and thus the main echoes can be reduced, thus increasing intelligibility. The DPAA emission pattern can be designed to reduce the energy reaching the live microphone connected to the DPAA input, thus better controlling unwanted acoustic feedback.
In crowd-control and military operations, the focusing and steering of the DPAA beam (without moving physically bulky loudspeakers and / or horns) produces a very strong sound field in remote areas. This sound field can be easily and quickly rearranged and easily delivered to the target by a tracking light source, resulting in a non-invasive but powerful sonic weapon. With a large array, or if a group of separate DPAA panels can be adjusted and placed at wide intervals, the focal region will have a much wider sound field than near the DPAA SET. It can be powerful (even at the bottom of the audible band, if the overall array size is large enough).
Any of the above aspects can be combined with the actual device to obtain the above-mentioned advantages.
<u style="single">A preferred embodiment of the first aspect of the present invention</u> Next, a preferred embodiment of the first aspect of the present invention will be described. This also utilizes the techniques of the other aspects described above, but will become apparent in due course.
Referring to FIG. 23, the digital sound projector 10 includes an array of transducers or loudspeakers 11 that are controlled to emit audio input signals as sound beams 12-1 and 12-2. , It can be sent in any direction in the half space in front of the array within the limited range. By utilizing a carefully selected reflection path, the listener 13 perceives the sound beam emitted from the array as if it originated from its last reflection position.
Figure 23 shows two sound beams 12-1 and 12-2. The first beam 12-1 is sent toward the side wall 161 which is a part of the room and is reflected directly toward the listener 13. The listener perceives this beam as coming from the reflection spot 17, and therefore from the right. The second beam 12-2 shown by the broken line reflects twice before reaching the listener 13. However, since the final reflex takes place in the rear corner, the listener perceives the sound as if it were emitted from a sound source behind him or her.
There are many possible uses for digital sound projectors, but the most effective one is to replace traditional surround sound systems with several separate loudspeakers located at different locations around the listener's location. That is. Digital sound projectors generate a beam for each channel of the surround sound audio signal and steer the beam in the right direction to listen without the need for many loudspeakers or extra wiring. You can create a true surround sound in the position of the person.
24 to 26 show the components of the digital sound projector in the form of a block diagram. At the input, the audio source material in a common format in pulse code modulation (PCM) form is a digital sound from devices such as compact discs (CDs), digital video discs (DVDs), etc. Received by the projector as an optical or coaxial digital data stream in S / PDIF format. However, other input digital data formats are also available. This input data can be a simple two-channel stereo pair, or a compressed and encoded multi-channel soundtrack such as Dolby Digital® or DTS®, or a large number of discrete digital digital sources of audio information. It can include any of the channels.
The coded and / or compressed multi-channel input is first decrypted and / or decompressed in the decoder using a device available for standard audio and video formats and licensed firmware. The firmware. It also incorporates an analog-to-digital converter (not shown) that can be connected (AUX) to an analog input source, which is immediately converted to a properly sampled digital format. The resulting output typically consists of 3 pairs, 4 pairs, or more channel pairs. In the field of surround sound, these channels are often referred to as left, right, center, surround (rear) left, and surround (rear) right channels. The signal may also have other channels, such as low frequency effect channels (LFEs).
Each of these channels or channel pairs is fed to a 2-channel sample rate converter [SRC] (or each channel can pass through a single channel SRC), resynchronized, and resampled. Do this to get an internal (or optionally external) standard sample rate clock [SSC] (typically about 48.8KHz or 97.6KHz) and a bit length (usually 24 bits) to get the internal system. Allow the clock to be independent of the source data clock. This sample rate conversion solves the problems of low clock speed accuracy, clock drift, and clock incompatibility. That is, if the final power output stage of the digital sound projector should be digital pulse width modulation [PWM] switching type for high efficiency, the digital data supplied to the PWM clock and PWM modulator. -It is desirable to have perfect synchronization with the clock. The SRC synchronizes this and isolates it from any external data clock fluctuations.
Finally, if two or more digital input channels have different data clocks (perhaps because they came from separate digital microphone systems, for example), again by SRC, all internally. Make sure to synchronize dissimilar signals. The output of the SRC is converted to 8 channels of 24-bit words at a sample rate generated internally at 48.8 KHz.
Data is processed using one or more (typically two or three) digital signal processor [DSP] units. These could be, for example, Texas Instruments' TMS320C6701 DSP running at 133MHz, which either does most of the computation in floating point format for ease of coding, or maximizes processing speed. Use fixed-point format to limit. Alternatively, digital signal processing can be performed in one or more field programmable gate array (FPGA) units, especially when performing fixed-point arithmetic. Yet another alternative is a hybrid of DSP and FPGA. Alternatively, some or all of the digital processing may be performed by customized silicon in the form of application specific integrated circuits (ASICs).
The DSP stage performs filtering of the digital audio data input signal to improve the equalization of frequency response and the frequency response (ie, transfer) of the acoustic output converter used in the final stage of the digital sound projector. Compensate for irregularities in the function).
Optionally, the number of channels to be processed separately is (preferably) one additively (one or more) low frequency effect [LFE] channels in this stage, or perhaps in a earlier or later processing stage. It is preferable to reduce the number by combining with the above other channels, for example, the central channel, and to minimize the processing after this stage. However, if a separate subwoofer must be used with the system, or if processing power is not an issue, more channels can be maintained throughout the processing chain.
The DSP stage also performs anti-alias and sound quality control filtering on all eight channels, eightx oversampling and interpolation for the entire 8x oversampling data rate, and 8 channels at 390KHz. Form a 24-bit word output sample of. The DSP also performs signal limiting and digital volume control.
The ARM microprocessor generates timing delay data for each converter from the real-time beam steering set value sent by the user to the digital sound projector by infrared remote control. Since the digital sound projector can independently steer each of the output channels (one steered output channel for each input channel), many delay calculations will be performed separately. This number is equal to the number of output channels multiplied by the number of transducers. Digital sound projectors can also dynamically steer each beam in real time, so calculations need to be quick. Once calculated, the delay request is distributed through the same parallel bus to the FPGA, where the delay is actually given to each stream of the digital data sample, as the digital data sample itself. The ARM core also handles all system initialization and external communications.
The signal stream is input to the Xilinx field programmable gate array logic. This logic controls a fast static buffered RAM device to generate the delay needed to feed each of the eight channels of digital audio data samples. One output converter (256 in this embodiment) produces a discrete delay of each channel.
Apodisation, or array aperture windowing (ie, graded weighting) Factors) is applied to the signal for each transducer as a function of the distance from the center of the array of each transducer to control the beam shape)) is applied separately to the delay signal version of each channel in the FPGA. .. Applying apodization here allows different output sound beams to have different beam shapes that are individually formed. These separately delayed and separately windowed digital sample streams, one for each of the eight channels and one for each of the 256 transducers, totaling 8 x 256 = 2048. It is a delayed version, summed up in the FPGA for each transducer, and produces a separate 390KHz 24-bit signal for each of the 256 transducer elements. Optionally, apodization or array aperture windowing should be performed for all channels at once after the addition stage for simplicity (instead of performing it separately for each channel before the addition stage). ) However, in this case, each sound beam output from the digital sound projector will have the same window function, which is not optimal.
The 24-bit and 390kHz 256 signals are then passed through a quantization / noise shaping circuit, also in the FPGA, to reduce the data sample word length to 8 bits at 390kHz, but in the audible band. Within (ie, the signal frequency band from ~ 20Hz to ~ 30Kz), the high signal-to-noise ratio [SNR] remains maintained.
One practically useful embodiment is to make the SSC an exact rational fraction of the DSP master processing clock speed. For example, 100MHz / 256 = 390,625Hz, which locks the sample data rate to the processing clock throughout the system. The digital PWM timing clock frequency is also advantageous if it is an exact rational fraction of the DSP master processing clock speed. In particular, the PWM clock frequency is an exact integer multiple of the internal digital audio sample data rate, eg 512 times the sample rate for 9-bit PWM (2).<sup>9</sup>(Because = 512) is advantageous. Reducing the digital data word length to 8 while increasing the sample rate at the same time is useful for several reasons:
i) By increasing the sample rate, the resolution of the data word delay can be increased. For example, at a data rate of 48 KHz, the minimum possible delay step size is one sample period, or ~ 21 microseconds, while at a 195 KHz data rate, the minimum possible delay step size is (1 sample period). ~ 5.1 microseconds. It is important to have a fine sound-path-length compensation resolution (= time delay resolution multiplied by the speed of sound) compared to the diameter of the acoustic output converter. During 21 microseconds, aerial sound propagates about 7 mm over NTP. This is too coarse resolution when using a transducer with a diameter of only 10 mm.
ii) Converting PCM data directly to a digital PWM with a practical clock speed is easier as the word length is shorter. For example, a 48KHz data rate and 16-bit length requires a PWM clock speed of 65536 x 48KHz to 3.15GHz (almost impractical), while a 195KHz data rate and 8-bit length requires 256x390KHz. A PWM clock speed of ~ 100MHz (just practical) is required.
iii) Increasing the sample rate increases the signal bandwidth obtained at half the sample rate, for example, at a sample rate of ~ 195 KHz, the signal bandwidth obtained is ~ 96 KHz. The quantization process (reduction of the number of bits) effectively adds quantization noise to the digital data and spectrally shapes the noise generated by the quantization process to base it in the region between it and the top edge of the baseband. Most can be moved to frequencies higher than the band signal (ie, above ~ 20KHz in this case). The effect is that almost all of the original signal information is carried in a digital data stream, with very little signal-to-noise ratio reduction here.
The data stream with reduced sample word width is distributed to 26 serial data streams, each at 31 Mb / s, and additional volume data. Each data stream is assigned to one of 26 driver boards. As shown in FIG. 25, the driver circuit boards are preferably physically close to the transducers they drive, but each of the transducers they control is provided with a pulse width modulation class-BD output driver circuit. In this example, each driver board is connected to 10 transducers, which allows the transducers to go directly to the output of the class-BD output driver circuit, without any intervention of a low pass filter [LPF}. Be connected.
Each PWM generator drives a class-D power switch or output stage that drives one converter directly, or a pair of adjacent converters connected in series or in parallel. Digitally adjusting the power supply to the Class-D power switch can control the output power level to the transducer. By controlling this power supply over a wide range, for example 10: 1, the power to the converter is over a much wider range, 100: 1 for a voltage range of 10: 1, or generally a voltage range of N: 1. Against N<sup>2</sup>It can be controlled by 1. In this way, a wide range of level control (or "volume" control) can be performed without shortening the digital word length, so that signal degradation due to further quantization (or resolution degradation) does not occur. To change the power supply, use a low-loss switching regulator mounted on the same printed circuit board (PCB) as the Class-D power switch. One switching regulator is used for each class-D switch to minimize intermodulation of power lines. Costs can be reduced by using each switching regulator for every two, three, four, or other integer multiples of Class-D power switches.
Class-D power switches or output stages drive the acoustic output transducer directly. Driving a normal Class-D power amplifier, i.e., a very commonly used so-called "Class-AD" amplifier, has an electronic low-pass filter [LPF] between the Class-D power stage and the transducer. ] (Always an analog electronic LPF) must be placed, which is a common form of magnetic transducers (even stronger with piezoelectric transducers), but with high energy at the output of class-AD amplifiers. This is because it provides a low load impedance with respect to the frequency. For example, a class-AD amplifier with a zero baseband input signal will have a continuous PWM switching frequency (in this case, ~ 50 or 100MHz) at its output. Generates a 1: 1 mark-spatial ratio [MSR] output signal with maximum amplitude (usually bipolar) at) and dissipates the maximum power obtained at this load when connected to a nominal 8 ohm load. No useful acoustic output signal is obtained. Commonly used electronic LPF cut-off frequencies are the highest desired signal output frequencies (eg>>. It is higher than 20KHz), but much lower than the PWM switching frequency (eg ~ 50MHz), so it effectively shuts off PWM carriers and minimizes power waste. Such LPFs must transfer maximum signal power to electrical loads (eg, acoustic transducers) while reducing power loss as much as possible. These LPFs typically use at least two power inductors and, or more often, three capacitors. LPFs are bulky and relatively expensive to configure. For single-channel (or several-channel) amplifiers, such LPFs can be tolerated for cost reasons, and more importantly, PWM amplifiers are housed separately from their load (eg, traditional loudspeakers). Such LPFs are needed if these loads must be connected with long leads in some cases for completely different reasons: high frequency PWM carriers enter the connecting leads. However, this is to prevent the possibility of generating unwanted stray electromagnetic radiation [EMI] with a relatively large amplitude from becoming very high.
In digital sound projectors, the acoustic transducers are directly connected to physically adjacent PWM power switches by short reeds, all housed in the same enclosure, so there are no EMI issues. In digital sound projectors, PWM generators are of the type known as class-BD, which generate class-DB PWM signals and drive output power switches. On the other hand, they drive an acoustic output transducer. Class-BD PWM output signals have the property of returning to zero during maximum amplitude bipolar pulse output, and are therefore three states, not two states like class-AD signals. Therefore, when the digital input signal to the class-BD PWM system is zero, the class-BD power output state is zero, not the maximum power bipolar 1: 1 MSR signal as produced by class-AD PWM. .. That is, class-BD The PWM power switch supplies zero power to the load (acoustic transducer) in this state. No LPF is needed as there is no full power PWM carrier signal to block. Therefore, in digital sound projectors, the need for an array of power LPFs is eliminated by directly driving the integrated array of transducers using an array of class-BD PWM amplifiers, resulting in a significant reduction in cost and power loss. Is achieved. Class-BDs are rarely used in traditional audio amplifiers because, first of all, it is more difficult to make a very linear Class-BD amplifier than a Class-AD amplifier with similar linearity. Yes, and secondly, for the reasons mentioned above, LPFs are generally required anyway in view of EMI, counteracting the major effects of Class-BD.
The acoustic output transducer itself is a very effective electroacoustic LPF, so absolutely minimal PWM carriers are emitted as acoustic energy from the Class-BD PWM stage. For this reason, digital array loudspeakers in digital sound projectors combine class-BD PWM with direct coupling to an acoustic transducer in the same box and do not use an electronic LPF to achieve high efficiency, high power, It is a very effective and cost-effective solution for driving multiple converters. Furthermore, for a digital sound projector, the sound of any one (or more) output channel that corresponds to one of the input channels that the listener hears is the sum of the sounds from each of the acoustic output transducers, and therefore. In relation to the sum of the outputs from each of the power amplification stages driving these transducers, the asymmetric error in the power switch and converter outputs averages zero and is almost inaudible. Therefore, the advantage of array loudspeakers configured as described above is that they are more tolerant of the quality of the individual components than in traditional non-array audio systems.
In a particular embodiment of a digital sound projector, 254 sound output transducers are arranged in a roughly rectangular range of triangular arrays with one axis of the array vertical (and from 20 transducers). The range consists of seven vertical columns, each separated by six columns of 19 converters), and every other two output converters in each vertical column of the converter are directly below the converter and electricity. When connected in series or in parallel, 132 different versions are obtained for each of the channels, in this example the number of channels is 5. That is, there are a total of 660 channels. Making the transducer diameter small enough to ensure radiation from the transducer in approximately all directions up to high audio frequencies (eg 12KHz to 15KHz) is a small angle for the digital sound projector. This is important if the beam of sound from the plane of the array must be steerable. That is, the diameter of the transducer is optimally between 5 mm and 30 mm for application over the entire audio band. The spacing between converters is small compared to the shortest wavelength of sound emitted by a digital sound projector, which is the "spurious" sidelobe of acoustic emission (ie, accidentally generated and emitted in the desired direction). It is desirable to minimize the generation of unsound energy beams). By actually examining the possible transducer sizes, it was found that the best transducer spacing was in the range of 5 mm to 45 mm. A triangular array layout is also best suited for mounting transducers in an array with high areal density.
As shown in FIG. 26, the digital sound projector user interface is mounted on any properly connected video display, such as a plasma screen, for on-screen display of configuration, status and control information. Generate overlay graphics. To do this, the video signal from any connected audio-visual source (eg, a DVD player) may pass through the digital sound projector on the way to the display screen, the digital sound projector. Status and command information is also superimposed on the program video. The process delay of the end-to-end signal processing operation of the digital sound projector is long enough (eg, the length of the compensating filter running on the first two DSPs that depends on the linearity of the transducer and the required equalization). (If long) To avoid lip sync issues, an optional video frame storage can be incorporated into the passing video path to resynchronize the displayed video with the output sound.
<figref num="1">The figure which shows the simple single input device.</figref><figref num="2">Block diagram of a multi-input device.</figref><figref num="3">Block diagram of a general-purpose distributor.</figref><figref num="4">Block diagram of a linear amplifier and a digital amplifier used in a preferred embodiment of the present invention.</figref><figref num="5">The figure which shows the interconnection of several arrays having a common control and input stage.</figref>
<figref num="6">The figure which shows the distributor by the 1st aspect of this invention.</figref><figref num="7A">The figure which shows one of four kinds of sound fields which can be obtained by using the apparatus of 1st Embodiment of this invention.</figref><figref num="7B">The figure which shows another one of four kinds of sound fields which can be obtained by using the apparatus of 1st Embodiment of this invention.</figref><figref num="7C">The figure which shows another one of four kinds of sound fields which can be obtained by using the apparatus of 1st Embodiment of this invention.</figref><figref num="7D">The figure which shows another one of four kinds of sound fields which can be obtained by using the apparatus of 1st Embodiment of this invention.</figref>
<figref num="8">Diagram showing three different beam paths obtained when three sound channels are directed in different directions indoors.</figref><figref num="9">The figure which shows the device which delays each channel to take into account the different propagation distances.</figref><figref num="10">The figure which shows the device which delays a video signal according to the delay given to an audio channel.</figref>
<figref num="11A">The figure which shows one of the various window functions used to describe the third aspect of this invention.</figref><figref num="11B">The figure which shows another one of the various window functions used to describe the third aspect of this invention.</figref><figref num="11C">The figure which shows another one of the various window functions used to describe the third aspect of this invention.</figref><figref num="11D">The figure which shows another one of the various window functions used to describe the third aspect of this invention.</figref>
<figref num="12">The figure which shows the apparatus which applies a different window function to a different channel.</figref><figref num="13">A block diagram showing a device capable of shaping different frequencies in different ways.</figref><figref num="14">The figure which shows the apparatus which derives a different frequency band to a separate output converter.</figref><figref num="15">The figure which shows the apparatus which derives a different frequency band into a set of overlapping output converters.</figref><figref num="16">The front view of the array is shown, and the symbols represent the frequency bands output by each converter.</figref>
<figref num="17">FIG. 5 shows an array of output transducers having a dense converter region near the center according to a fourth aspect of the present invention.</figref><figref num="18">The figure which shows the single converter which has an elongated structure.</figref><figref num="19">The figure which shows the array of the converter shown in FIG.</figref><figref num="20">Top view showing an array of output transducers and a reflective / resonant screen for surround sound effects.</figref><figref num="21">Top view showing the transducer array and the reflected / resonant surface, as well as the beam pattern reflected on the surface.</figref>
<figref num="22">A side view showing an array to which a video camera is attached according to a seventh aspect of the present invention.</figref><figref num="23">The figure which shows the typical setting of the loudspeaker system by 1st Embodiment of this invention.</figref><figref num="24">A block diagram of a first portion of a digital loudspeaker system according to a preferred embodiment of the first aspect of the present invention.</figref><figref num="25">A block diagram of a second part of a digital loudspeaker system according to a preferred embodiment of the first aspect of the present invention.</figref><figref num="26">A block diagram of a third part of a digital loudspeaker system according to a preferred embodiment of the first aspect of the present invention.</figref>
Every citation, both ways
| Document | Relation | Office |
|---|---|---|
| JP6205496A | Cites | Japan |
| JP9233591A | Cites | Japan |
| JP4132499A | Cites | Japan |
| JP4127700A | Cites | Japan |
| JP213097A | Cites | Japan |
| JP8181962A | Cites | Japan |
| JP6178379A | Cites | Japan |
| WO0123104A1 | Cites | World Intellectual Property Organization (WIPO) |
| JP2000505627A | Cites | Japan |
23 members in 8 offices
Priority claims14
| Document | Office | Kind | Date |
|---|---|---|---|
| 0107699 | United Kingdom | A | |
| 0107699 | United Kingdom | A | |
| 01076991 | United Kingdom | – | |
| 0200291 | United Kingdom | A | |
| 0200291 | United Kingdom | A | |
| 02002913 | United Kingdom | – | |
| 0201472 | United Kingdom | W | |
| 0201472 | United Kingdom | W | |
| 2001200107699 | – | – | – |
| 2002200200291 | – | – | – |
| 2002001472 | – | – | – |
| GB20010007699 | – | – | – |
| GB20020000291 | – | – | – |
| WO2002GB01472 | – | – | – |
Members23
| Document | Office | Kind | |
|---|---|---|---|
| GB2373956A | United Kingdom | A | |
| WO02078388A2 | World Intellectual Property Organization (WIPO) | A2 | |
| AU2002244845A1 | Australia | A1 | |
| GB2376595A | United Kingdom | A | |
| WO03059005A2 | World Intellectual Property Organization (WIPO) | A2 | |
| AU2003201640A1 | Australia | A1 | |
| AU2003201640A8 | Australia | A8 | |
| WO03059005A3 | World Intellectual Property Organization (WIPO) | A3 | |
| GB2376595B | United Kingdom | B | |
| WO02078388A3 | World Intellectual Property Organization (WIPO) | A3 | |
| KR20040004566A | Republic of Korea | A | |
| EP1402755A2 | European Patent Office (EPO) | A2 | |
| US2004151325A1 | United States of America | A1 | |
| GB2398959A | United Kingdom | A | |
| JP2004531125A | Japan | A | |
| CN1605225A | China | A | |
| JP2007236005A | Japan | A | |
| US7515719B2 | United States of America | B2 | |
| US2009161880A1 | United States of America | A1 | |
| CN100539737C | China | C | |
| KR100922910B1 | Republic of Korea | B1 | |
| CN101674512A | China | A | |
| JP4445705B2This record | Japan | B2 |
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Numbers
- Publication
- 4445705
- Publication, DOCDB
- 4445705
- Publication, EPODOC
- JP4445705B
- Application
- 576475
- Application, DOCDB
- 2002576475
- Application, EPODOC
- JP20020576475
Titles2
- Japanese
- 音場を作り出す方法および装置
- English
- Methods and devices for creating sound fields
Classification
- CPC, 11
- F41H13/0081
- H04S3/00
- G10K15/04
- H04R1/26
- H04R1/403
- H04R3/12
- H04R2201/401
- H04R2203/12
- H04R2205/022
- H04S1/002
- H04S3/002
- IPC, 8
- H04S7 00
- H04R1 40
- H04S3 00
- H04N5 232
- G10K15 04
- H04R1 26
- H04R3 12
- H04S1 00