Patent Publication Number: US-11659344-B2

Title: Sound signal processing method, sound signal processing device, and storage medium that stores sound signal processing program

Description:
CROSS REFERENCE TO RELATED APPLICATIONS 
     This Nonprovisional application claims priority under 35 U.S.C. § 119(a) on Patent Application No. 2020-096756 filed in Japan on Jun. 3, 2020, the entire contents of which are hereby incorporated by reference. 
     BACKGROUND 
     Technical Field 
     One embodiment of the present disclosure relates to a sound signal processing method for processing an obtained sound signal. 
     Background Information 
     In facilities such as a concert hall, various genres of music are played and speeches such as a lecture are given. Such facilities are required to have various acoustic characteristics (for example, reverberant characteristics). For example, a performance requires a relatively long reverberant, and a speech requires a relatively short reverberant. 
     However, in order to physically change the reverberant characteristics in a hall, it is necessary to change the size of an acoustic space by, for example, moving the ceiling, which requires a very large-scale facility. 
     In view of the above, for example, a sound field control device as shown in JP H06-284493 A performs processing that supports a sound field by generating a reverberant sound by processing the sound obtained by a microphone with a finite impulse response (FIR) filter, and outputting the reverberant sound from a speaker installed in a hall. 
     SUMMARY 
     However, the sense of localization is blurred only by adding a reverberant sound. Recently, it has been desired to realize clearer sound image localization and richer space expansion. 
     In view of the above, an object of one embodiment of the present disclosure is to provide a sound signal processing method for controlling a richer acoustic space. 
     The sound signal processing method includes receiving a sound signal, generating an early reflection sound control signal that reproduces an early reflection sound and a reverberant sound control signal that reproduces a reverberant sound from the sound signal, controlling volume of the sound signal and distributing the sound signal to generate a direct sound control signal that reproduces a direct sound, and mixing the direct sound control signal, the early reflection sound control signal, and the reverberant sound control signal to generate an output signal. 
     The sound signal processing method can realize clearer sound image localization and richer space expansion. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG.  1    is a transparent perspective view schematically illustrating a space of a first embodiment; 
         FIG.  2    is a block diagram illustrating a configuration of a sound field support system of the first embodiment; 
         FIG.  3    is a flowchart showing operation of a sound signal processing device; 
         FIG.  4    is a plan view schematically illustrating a relationship between a room  62 , speakers  51 A to  51 J, and a sound source  65 ; 
         FIG.  5 A  is a schematic diagram illustrating an example of classification of sound types in a time waveform of an impulse response used for a filter coefficient; 
         FIG.  5 B  is a schematic diagram illustrating a time waveform of a filter coefficient set to an FIR filter  24 A; 
         FIG.  6 A  is a schematic diagram illustrating an impulse response set to an FIR filter  24 B; 
         FIG.  6 B  is a schematic diagram illustrating a time waveform of a filter coefficient set to the FIR filter  24 B; and 
         FIG.  7    is a plan view schematically illustrating a relationship between a space  620  and the room  62 . 
     
    
    
     DETAILED DESCRIPTION 
       FIG.  1    is a transparent perspective view schematically illustrating a room  62  constituting a space.  FIG.  2    is a block diagram illustrating a configuration of a sound field support system  1 . 
     The room  62  constitutes a space having a substantially rectangular parallelepiped shape. A sound source  65  exists on a stage  60  at the front of the room  62 . The back of the room  62  corresponds to audience seats where listeners sit. The sound source  65  is, for example, a voice, a singing sound, an acoustic musical instrument, an electric musical instrument, an electronic musical instrument, or the like. 
     The shape of the room  62 , the arrangement of sound sources, and the like are not limited to the example shown in  FIG.  1   . A sound signal processing method and a sound signal processing device of the present disclosure can provide a desired sound field in a space of any shape, and can realize a richer sound image and richer space expansion than before. 
     The sound field support system  1  includes, in the room  62 , a directional microphone  11 A, a directional microphone  11 B, a directional microphone  11 C, an omnidirectional microphone  12 A, an omnidirectional microphone  12 B, an omnidirectional microphone  12 C, speakers  51 A to  51 J, and speakers  61 A to  61 F. 
     The speakers  51 A to  51 J are set on a wall surface. The speakers  51 A to  51 J are speakers with relatively high directivity, and mainly output sound toward audience seats. The speakers  51 A to  51 J output an early reflection sound control signal that reproduces an early reflection sound. Further, the speakers  51 A to  51 J output a direct sound control signal that reproduces a direct sound of the sound source. 
     The speakers  61 A to  61 F are installed on the ceiling. The speakers  61 A to  61 F are speakers with relatively low directivity, and output sound to the entire room  62 . The speakers  61 A to  61 F output a reverberant sound control signal that reproduces a reverberant sound. Further, the speakers  61 A to  61 F output a direct sound control signal that reproduces a direct sound of the sound source. The number of speakers is not limited to the number shown in  FIG.  1   . 
     The directional microphone  11 A, the directional microphone  11 B, and the directional microphone  11 C mainly collect the sound of the sound source  65  on the stage. However, as shown in  FIG.  2   , the sound of the sound source  65  may be received by line input. In the line input, the sound output from a sound source of an instrument or the like is not collected with a microphone and receive, but a sound signal is received from an audio cable or the like connected to the sound source. Alternatively, as the sound of the sound source  65 , the voice or singing sound of a performer such as a speaker or a singer may be received from a hand microphone, a stand microphone, a pin microphone, or the like. The sound of the sound source is preferably collected at a high SN ratio. 
     The omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C are installed on the ceiling. The omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C collect the entire sound in the room  62 , including the direct sound of the sound source  65  and the reflection sound in the room  62 . The number of the directional microphones and the omnidirectional microphones shown in  FIG.  1    is three each. However, the number of microphones is not limited to that in the example shown in  FIG.  1   . Further, the installation positions of the microphone and the speaker are not limited to those in the example shown in  FIG.  1   . 
     In  FIG.  2   , the sound field support system  1  includes a sound signal processor  10  and a memory  31  in addition to the configuration shown in  FIG.  1   . The sound signal processor  10  is mainly composed of a CPU and a digital signal processor (DSP). The sound signal processor  10  functionally includes a sound signal obtainer  21 , a gain adjuster  22 , a mixer  23 , a panning processor  23 D, a finite impulse response (FIR) filter  24 A, an FIR filter  24 B, a level setter  25 A, a level setter  25 B, an output signal generator  26 , an output unit  27 , a delay adjuster  28 , a position information obtainer  29 , an impulse response obtainer  151 , and a level balance adjuster  152 . The sound signal processor  10  is an example of the sound signal processing device of the present disclosure. 
     The CPU constituting the sound signal processor  10  reads an operation program stored in the memory  31  and controls each configuration. The CPU functionally constitutes the position information obtainer  29 , the impulse response obtainer  151 , and the level balance adjuster  152  by the operation program. The operation program does not need to be stored in the memory  31 . The CPU may download the operation program from, for example, a server (not shown) each time. 
       FIG.  3    is a flowchart showing the operation of the sound signal processor  10 . First, the sound signal obtainer  21  obtains a sound signal (S 11 ). The sound signal obtainer  21  obtains a sound signal from the directional microphone  11 A, the directional microphone  11 B, the directional microphone  11 C, the omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C. Alternatively, the sound signal obtainer  21  may receive a sound signal from an electric musical instrument, an electronic musical instrument, or the like by line input. Further, the sound signal obtainer  21  may receive a sound signal from a microphone installed directly on a musical instrument or the performer such as a pin microphone. When obtaining an analog signal, the sound signal obtainer  21  converts the analog signal into a digital signal and outputs the signal. 
     The gain adjuster  22  adjusts a gain of the sound signal obtained by the sound signal obtainer  21 . The gain adjuster  22  sets, for example, a gain of the directional microphone at a position close to the sound source  65  to be high. The gain adjuster  22  is not an essential configuration in the present disclosure. 
     The mixer  23  mixes the sound signal whose gain is adjusted by the gain adjuster  22 . Further, the mixer  23  distributes the mixed sound signal to a plurality of signal processing systems. 
     The mixer  23  outputs the distributed sound signal to the panning processor  23 D, the FIR filter  24 A, and the FIR filter  24 B. 
     For example, the mixer  23  distributes the sound signals obtained from the directional microphone  11 A, the directional microphone  11 B, and the directional microphone  11 C to ten signal processing systems according to the speakers  51 A to  51 J. Alternatively, the mixer  23  may distribute line-inputted sound signals to ten signal processing systems according to the speakers  51 A to  51 J. 
     Further, the mixer  23  distributes the sound signals obtained from the omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C to six signal processing systems according to the speakers  61 A to  61 F. 
     The mixer  23  outputs the sound signals mixed in the ten signal processing systems to the panning processor  23 D and the FIR filter  24 A. Further, the mixer  23  outputs the sound signals mixed in six signal processing systems to the FIR filter  24 B. 
     Hereinafter, the six signal processing systems that output a sound signal to the FIR filter  24 B will be referred to as a first system or a reverberant sound system, and the ten signal processing systems that output a sound signal to the FIR filter  24 A will be referred to as a second system or an early reflection sound system. Further, ten signal processing systems that output a sound signal to the panning processor  23 D will be referred to as a third system or a direct sound system. The FIR filter  24 A corresponds to an early reflection sound control signal generator, and the FIR filter  24 B corresponds to a reverberant sound control signal generator. The panning processor  23 D corresponds to a direct sound control signal generator. 
     The mode of distribution is not limited to that in the above example. For example, sound signals obtained from the omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C may be distributed to the direct sound system or the early reflection sound system. Further, a line-inputted sound signal may be distributed to the reverberant sound system. Further, a line-inputted sound signal and sound signals obtained from the omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C may be mixed and distributed to the direct sound system or the early reflection sound system. 
     The mixer  23  may have a function of electronic microphone rotator (EMR). The EMR is a method of flattening the frequency characteristics of a feedback loop by changing a transfer function between a fixed microphone and a speaker over time. The EMR is a function that switches a connection relationship between a microphone and a signal processing system from moment to moment. The mixer  23  outputs a sound signal obtained from the directional microphone  11 A, the directional microphone  11 B, and the directional microphone  11 C to the panning processor  23 D and the FIR filter  24 A by switching an output destination. Alternatively, the mixer  23  outputs the sound signal obtained from the omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C to the FIR filter  24 B by switching an output destination. In this manner, the mixer  23  can flatten the frequency characteristics of an acoustic feedback system from a speaker to a microphone in the room  62 . Further, the mixer  23  can ensure stability against howling. 
     Next, the panning processor  23 D controls the volume of each sound signal of the direct sound system according to the position of the sound source  65  (S 12 ). In this manner, the panning processor  23 D generates a direct sound control signal. 
       FIG.  4    is a plan view schematically illustrating a relationship between the room  62 , the speakers  51 A to  51 J, and the sound source  65 , In the example of  FIG.  4   , the sound source  65  is located on the right side of the stage when viewed from the audience seats. The panning processor  23 D controls the volume of each sound signal of the direct sound system so that the sound image is localized at the position of the sound source  65 . 
     The panning processor  23 D obtains the position information of the sound source  65  from the position information obtainer  29 . The position information is information indicating two-dimensional or three-dimensional coordinates with respect to a certain position of the room  62 . The position information of the sound source  65  can be obtained by a beacon and a tag that transmit and receive a radio wave of, for example, Bluetooth (registered trademark). In the room  62 , at least three beacons are installed in advance. The sound source  65  includes a tag. That is, a tag is attached to a performer or an instrument. Each beacon transmits and receives radio waves to and from the tag. Each beacon measures the distance to the tag based on the time difference between transmitting and receiving of radio waves. If the position information obtainer  29  obtains the position information of the beacon in advance, the position of the tag can be uniquely obtained by measurement of the distances from at least three beacons to the tag. 
     The position information obtainer  29  obtains the position information of the sound source  65  by obtaining the position information of the tag measured in the above manner. Further, the position information obtainer  29  obtains the position information of each of the speakers  51 A to  51 J and the speakers  61 A to  61 F in advance. 
     The panning processor  23 D controls the volume of each sound signal output to the speakers  51 A to  51 J and the speakers  61 A to  61 F so that the sound image is localized at the position of the sound source  65  based on the obtained position information and the position information of the speakers  51 A to  51 J and the speakers  61 A to  61 F, so as to generate the direct sound control signal. 
     The panning processor  23 D controls the volume according to the distance between the sound source  65  and each of the speakers, the speakers  51 A to  51 J and the speakers  61 A to  61 F. For example, the panning processor  23 D increases the volume of the sound signal output to the speaker near the position of the sound source  65 , and decreases the volume of the sound signal output to the speaker far from the position of the sound source  65 . In this manner, the panning processor  23 D can localize the sound image of the sound source  65  at a predetermined position. For example, in the example of  FIG.  4   , the panning processor  23 D increases the volume of the sound signal output to the three speakers  51 F,  51 G, and  51 H close to the sound source  65 , and decreases the volume of the other speakers. In this manner, the sound image of the sound source  65  is localized on the right side of the stage when viewed from the audience seats. 
     If the sound source  65  moves to the left side of the stage, the panning processor  23 D changes the volume of each sound signal output to the speakers  51 A to  51 J and the speakers  61 A to  61 F based on the position information of the moved sound source  65 . For example, the panning processor  23 D increases the volume of the sound signal output to the speakers  51 A,  51 B, and  51 F, and decreases the volume of the other speakers. In this manner, the sound image of the sound source  65  is localized on the left side of the stage when viewed from the audience seats. 
     As described above, the sound signal processor  10  realizes a distribution processor of the present disclosure by the mixer  23  and the panning processor  23 D. 
     Next, the FIR filter  24 A and the FIR filter  24 B perform indirect sound generation processing (S 13 ). The indirect sound generation processing is processing of individually generating an early reflection sound control signal that reproduces an early reflection sound and a reverberant sound control signal that reproduces a reverberant sound. The FIR filter  24 A and the FIR filter  24 B correspond to an indirect sound generator of the present disclosure. 
     First, the impulse response obtainer  151  sets filter coefficients of the FIR filter  24 A and the FIR filter  24 B. Here, impulse response data set to the filter coefficient will be described.  FIG.  5 A  is a schematic diagram illustrating an example of classification of sound types in a time waveform of an impulse response used for the filter coefficient, and  FIG.  5 B  is a schematic diagram illustrating a time waveform of the filter coefficient set to the FIR filter  24 A.  FIGS.  6 A and  6 B  are schematic diagrams illustrating time waveforms of the filter coefficient set in the FIR filter  24 B. 
     As shown in  FIG.  5 A , an impulse response can be classified into a direct sound, an early reflection sound, and a reverberant sound arranged on the time axis. Then, as shown in  FIG.  5 B , the filter coefficient set to the FIR filter  24 A is set by the portion of the early reflection sound excluding the direct sound and the reverberant sound in the impulse response. As shown in  FIG.  6 A , the filter coefficient set to the FIR filter  24 B is set by the reverberant sound excluding the direct sound and the early reflection sound in the impulse response. As shown in  FIG.  6 B , the FIR filter  24 B may be set by the early reflection sound and the reverberant sound excluding the direct sound in the impulse response. 
     Data of the impulse response is stored in the memory  31 . The impulse response obtainer  151  obtains data of an impulse response from the memory  31 . However, the data of an impulse response does not need to be stored in the memory  31 . The impulse response obtainer  151  may download the data of an impulse response from, for example, a server (not shown) each time. 
     The impulse response obtainer  151  may obtain the data of an impulse response in which only the early reflection sound is cut out in advance and set the data to the FIR filter  24 A. Alternatively, the impulse response obtainer  151  may obtain the data of an impulse response including the direct sound, the early reflection sound, and the reverberant sound, cut out only the early reflection sound, and set the data to the FIR filter  24 A. Similarly, when only the reverberant sound is used, the impulse response obtainer  151  may obtain the data of an impulse response obtained by cutting out only the reverberant sound in advance and set the data to the FIR filter  24 B. Alternatively, the impulse response obtainer  151  may obtain the data of an impulse response including the direct sound, the early reflection sound, and the reverberant sound, cut out only the reverberant sound, and set the data to the FIR filter  24 B. 
       FIG.  7    is a plan view schematically illustrating a relationship between a space  620  and the room  62 . As shown in  FIG.  7   , the data of an impulse response is measured in advance in the predetermined space  620  such as a concert hall or a church where the sound field is to be reproduced. For example, the data of an impulse response is measured in a manner that a test sound (pulse sound) is generated at the position of the sound source  65  and the sound is collected with a microphone. 
     The data of an impulse response may be obtained at any position in the space  620 . However, it is preferable to measure the data of an impulse response of the early reflection sound using a directional microphone installed near a wall surface. The early reflection sound is a clear reflection sound in the direction of arrival. Therefore, by measuring the data of an impulse response with a directional microphone installed near a wall surface, it is possible to precisely obtain the reflection sound data of the target space. In contrast, the reverberant sound is a reflection sound in which the direction of arrival of the sound is uncertain. Therefore, the data of an impulse response of the reverberant sound may be measured with a directional microphone installed near the wall surface, or may be measured by using an omnidirectional microphone different from one used for the early reflection sound. 
     The FIR filter  24 A convolves data of different impulse responses with ten sound signals of the second system. When there are a plurality of signal processing systems, the FIR filter  24 A and the FIR filter  24 B may be provided for each of the signal processing systems. For example, ten of the FIR filters  24 A may be provided. 
     As described above, when a directional microphone installed near a wall surface is used, the data of an impulse response is measured with a separate directional microphone for each signal processing system. For example, as shown in  FIG.  7   , for a signal processing system corresponding to the speaker  51 J installed on the right rear side from the stage  60 , the data of an impulse response is measured with a directional microphone  510 J installed near a wall surface on the right rear side from the stage  60 . 
     The FIR filter  24 A convolves the data of an impulse response with each sound signal of the second system. The FIR filter  24 B convolves the data of an impulse response with each sound signal of the first system. 
     The FIR filter  24 A generates an early reflection sound control signal that reproduces an early reflection sound in a predetermined space by convolving the data of an impulse response of a set early reflection sound with an input sound signal. The FIR filter  24 B generates a reverberant sound control signal that reproduces a reverberant sound in a predetermined space by convolving the data of an impulse response of a set reverberant sound with an input sound signal. 
     The level setter  25 A adjusts the level of the early reflection sound control signal. The level setter  25 B adjusts the level of the reverberant sound control signal. The level balance adjuster  152  sets a level adjustment amount of the panning processor  23 D, the level setter  25 A, and the level setter  25 B. The level balance adjuster  152  refers to a level of each of the direct sound control signal, the early reflection sound control signal, and the reverberant sound control signal, and adjusts the level balance of these signals. For example, the level balance adjuster  152  adjusts the level balance between the last level in time of the direct sound control signal and the first component in time of the early reflection sound control signal. Further, the level balance adjuster  152  adjusts the balance between the level of the last component in time of the early reflection sound control signal and the level of the first component in time of the reverberant sound control signal. Alternatively, the level balance adjuster  152  may adjust the balance between the power of a plurality of components of the latter half in time of the early reflection sound control signal and the power of a component of the first half in time of the reverberant sound control signal. In this manner, the level balance adjuster  152  can individually control the sounds of the direct sound control signal, the early reflection sound control signal, and the reverberant sound control signal, and control the sounds to be in an appropriate balance according to the space to which the sounds are applied. 
     The delay adjuster  28  adjusts delay time according to the distance between an optional microphone and a plurality of speakers. For example, for a plurality of speakers, the delay adjuster  28  sets the delay time to be smaller in ascending order of distances between the directional microphone  11 B and the speakers. Alternatively, the delay adjuster  28  adjusts the delay time from the positions of the sound source and the microphone for which to measure the impulse response in the space  620  that reproduces the sound field. For example, in the FIR filter  24 A, when the impulse response data measured at the directional microphone  510 J installed in the space  620  is convolved with the speaker  51 J, the delay time corresponding to the distance between the directional microphone  510 J and the sound source  65  in the space  620  is set to the delay time of the speaker  51 J in the delay adjuster  28 . In this manner, the early reflection sound control signal and the reverberant sound control signal reach the listener later than the direct sound control signal, so that clear sound image localization and rich space expansion are realized. 
     It is preferable that the sound signal processor  10  do not perform delay adjustment on the direct sound control signal. If the position of the sound source  65  changes significantly in a short period of time when the sound image localization is controlled by the delay adjustment, phase interference occurs between the sounds output from a plurality of speakers. By not performing the delay adjustment on the direct sound control signal, the sound signal processor  10  can maintain the timbre of the sound source  65  without causing phase interference even if the position of the sound source  65  changes significantly in a short time. 
     Next, the output signal generator  26  mixes the direct sound control signal, the early reflection sound control signal, and the reverberant sound control signal to generate an output signal (S 14 ). The output signal generator  26  may perform gain adjustment of each signal, adjustment of the frequency characteristics, and the like at the time of mixing. 
     The output unit  27  converts an output signal output from the output signal generator  26  into an analog signal. Further, the output unit  27  amplifies the analog signal. The output unit  27  outputs the amplified analog signal to a corresponding speaker (S 15 ). 
     With the above configuration, the sound signal processor  10  obtains a sound signal, controls the volume of the sound signal and distributes the sound signal to a plurality of systems, generates a direct sound control signal, an early reflection sound control signal, and a reverberant sound control signal from the sound signal, and mixes the distributed sound signal, the direct sound control signal, the early reflection sound control signal, and the reverberant sound control signal to generate an output signal. In this manner, the sound signal processor  10  realizes clearer sound image localization and richer space expansion than before. 
     In particular, the sound signal processor  10  realizes the localization of a sound source by controlling the volume of a sound signal distributed to a plurality of speakers based on the position information of the sound source. Accordingly, it is possible to uniformly localize a clear sound image over a wide range in real time without depending on a reproduction environment such as the number and arrangement of speakers. 
     Further, the sound signal processor  10  outputs an early reflection sound control signal and a reverberant sound control signal from a plurality of speakers in addition to a direct sound control signal. In this manner, the audience listens to the early reflection sound control signal and the reverberant sound control signal in addition to the direct sound control signal. Therefore, the audience does not pay attention only to a specific speaker to which the direct sound control signal is output. Therefore, even when the number of speakers is small and the distance between the speakers is wide, the sound image is not localized only in a specific speaker. 
     The omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C collect the entire sound in the room  62 , including the direct sound of the sound source  65  and the reflection sound in the room  62 . Therefore, if the sound signal processor  10  generates a reverberant sound control signal using sound signals obtained by the omnidirectional microphone  12 A, the omnidirectional microphone  12 B, and the omnidirectional microphone  12 C, the same reverberant is reproduced in the sound of the stage and in the sound of the audience seat. Therefore, for example, the same reverberant is reproduced in the sound of the performer and in the sound of the applause of the audience, and the audience can obtain a sense of unity. 
     Further, the early reflection sound has a smaller number of reflections than the reverberant sound that undergoes multiple reflections in the space. For this reason, the energy of the early reflection sound is higher than the energy of the reverberant sound. Therefore, by increasing the level of each speaker that outputs the early reflection sound control signal, the effect of the subjective impression of the early reflection sound can be improved, and the controllability of the early reflection sound can be improved. 
     Further, by reducing the number of speakers that output the early reflection sound control signal, it is possible to suppress an excessive increase in diffused sound energy. That is, the extension of the reverberant in the room due to the early reflection sound control signal can be suppressed, and the controllability of the early reflection sound can be improved. 
     As the speaker that outputs the direct sound control signal and the early reflection sound control signal is installed on the side of the room, which is located close to the audience, the speaker can be easily controlled to deliver a direct sound and an early reflection sound to the audience, and the controllability of the early reflection sound can be improved. Further, by installing the speaker that outputs the reverberant sound control signal on the ceiling of the room, it is possible to suppress the difference in the reverberant sounds due to the positions of the audience. 
     The description of the present embodiment is exemplary in all respects and is not restrictive. The scope of the present disclosure is shown not by the above-described embodiment but by the scope of claims. Furthermore, the scope of the present disclosure is intended to include all modifications within the meaning and scope equivalent to those of the claims.