阅读说明：本技术 一种用于会议系统的扬声器信号馈给方法 (Loudspeaker signal feeding method for conference system ) 是由 王恒 曾维坚 东莲正 李子强 陈科壬 高韦涵 朱镇熙 于 2019-10-18 设计创作，主要内容包括：本发明公开了一种用于会议系统的扬声器信号馈给方法,包括：获取目标区域的空间参数并构建三维坐标系；测量目标区域内听众席和发言席的混响时间值结合目标区域的空间参数计算得到目标区域常数值,算得目标区域内能布置扬声器的位置与听众席的最远距离值；将多个扬声器按规则布置在目标区域内；计算各个听众席到发言席的距离,扬声器到发言席的距离,结合音速计算得到信号延时值；获取目标区域内各扬声器的灵敏度级和指向性系数两个参数数据；根据扬声器的参数数据、听众席位置坐标、各扬声器位置坐标和目标区域常数值,计算得到信号增益值；将信号延时值和信号增益值上传至会议系统,对已按规则布置的扬声器进行对应的信号延时和增益进行配置。(The invention discloses a loudspeaker signal feeding method for a conference system, which comprises the following steps: acquiring space parameters of a target area and constructing a three-dimensional coordinate system; measuring reverberation time values of audition seats and speech seats in a target area, and combining the reverberation time values with space parameters of the target area to calculate constant values of the target area, and calculating the farthest distance value between the position where a loudspeaker can be arranged in the target area and the audition seats; arranging a plurality of loudspeakers in the target area according to a rule; calculating the distance from each auditorium to the speech seat, the distance from the loudspeaker to the speech seat, and calculating a signal delay value by combining the sound speed; acquiring two parameter data of sensitivity level and directivity coefficient of each loudspeaker in a target area; calculating to obtain a signal gain value according to the parameter data of the loudspeakers, the position coordinates of the auditorium, the position coordinates of each loudspeaker and the constant value of the target area; and uploading the signal delay value and the signal gain value to a conference system, and configuring corresponding signal delay and gain for the loudspeakers which are arranged according to the rule.)

1. A method of speaker signal feeding for a conferencing system, comprising:

acquiring space parameters of a target area and constructing a three-dimensional coordinate system;

measuring reverberation time values of the auditorium and the speaking seat in the target area, calculating to obtain a constant value of the target area according to the reverberation time values and the spatial parameters of the target area, and calculating to obtain a farthest distance value between a loudspeaker and the auditorium in the target area according to the constant value of the target area;

within the farthest distance value range, arranging a plurality of loudspeakers in the target area according to a rule and enabling the front axial directions of the loudspeakers to face an auditorium;

acquiring coordinate data of a speech seat position, coordinate data of each auditorium position and coordinate data of positions of all loudspeakers in the three-dimensional coordinate system, and calculating a signal delay value of each loudspeaker according to the coordinate data of the speech seat position, the coordinate data of each auditorium position and the coordinate data of the positions of all loudspeakers in combination with a sound propagation speed;

acquiring parameter data of each loudspeaker in a target area;

calculating to obtain a signal gain value corresponding to each loudspeaker according to the parameter data of the loudspeaker, the coordinate data of each auditorium position, the coordinate data of the position of each loudspeaker and the constant value of the target area;

and uploading the signal delay value and the signal gain value to a conference system, and configuring corresponding signal delay and gain parameters of the regularly arranged loudspeakers.

2. The method of claim 1, wherein measuring reverberation time values of an auditorium and a floor in the target region, calculating a target region constant value from the reverberation time values in combination with spatial parameters of the target region, and calculating a farthest distance value between a speaker and an auditorium in the target region from the target region constant value, comprises:

uniformly selecting a plurality of measuring points in the auditorium, measuring to obtain the reverberation time of each measuring point, and calculating to obtain the reverberation time average value according to the reverberation time of each measuring point;

calculating to obtain an average sound absorption coefficient value of the target area according to the reverberation time average value and the space parameter of the target area;

calculating to obtain a target area constant value according to the average sound absorption coefficient value, the reverberation time average value and the space parameter of the target area;

and calculating the farthest distance value between the loudspeaker and the auditorium in the target area according to the constant value of the target area.

3. The speaker signal feed method for a conferencing system of claim 1, wherein the regularly arranging a plurality of speakers within a target area comprises:

arranging the loudspeakers around the auditorium in a ring shape, wherein the horizontal distance between each loudspeaker and the auditorium closest to the loudspeaker is more than 1 meter, and the height from the ground is more than 1.2 meters;

the maximum horizontal distance between every two adjacent loudspeakers is 2 times of the distance between the corresponding two adjacent seats.

4. The method of claim 1, wherein said calculating a speaker's signal delay value from said floor location coordinate data, each floor location coordinate data, and each speaker location coordinate data in combination with sound propagation speed comprises:

calculating the sound propagation time from each position of the auditorium to the auditorium according to the coordinate data of the auditorium position and the coordinate data of each auditorium position and combining the sound propagation speed;

sequencing the sound propagation time from each position of the auditorium to the speech seat to obtain the auditorium position corresponding to the longest time value and the auditorium position corresponding to the shortest time value;

calculating the sound propagation time from each loudspeaker to the speech place according to the coordinate data of the speech place position and the coordinate data of the position of each loudspeaker and the sound propagation speed;

sequencing the sound propagation time from each loudspeaker to the speech position to obtain the loudspeaker position corresponding to the longest time value and the loudspeaker position corresponding to the shortest time value;

calculating the maximum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the longest sound propagation time and the position of the loudspeaker with the longest sound propagation time, the maximum sound propagation time value from the speech seat to the auditorium and the sound propagation speed;

calculating the minimum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the shortest sound propagation time and the position of the loudspeaker with the shortest sound propagation time, the minimum sound propagation time value from the speech seat to the auditorium and the sound propagation speed; sequencing the distance values from the loudspeakers to the speech seats to obtain a maximum loudspeaker distance value and a minimum loudspeaker distance value;

and calculating respective signal delay values corresponding to the rest of the loudspeakers according to the maximum loudspeaker distance value, the minimum loudspeaker distance value, the maximum signal delay value, the minimum signal delay value and the distance value from each loudspeaker to the speech seat.

5. The speaker signal feed method for a conferencing system of claim 1, wherein the parameter data for the speaker comprises: the characteristic sensitivity level of the loudspeaker and the directivity coefficient of the axial included angle of the loudspeaker.

6. The method of claim 1, wherein the calculating a signal gain value corresponding to each speaker based on the parameter data of the speaker, the coordinate data of each auditorium position, the coordinate data of each speaker position, and the target area constant value comprises:

calculating additional sound pressure levels generated by the loudspeakers at the positions of the auditoriums according to the parameter data of the loudspeakers, the coordinate data of the positions of the auditoriums, the coordinate data of the positions of the loudspeakers and the constant value of the target area;

calculating the reference sound pressure level of each position of the auditorium according to the parameter data of the loudspeaker, and calculating the sound pressure level difference value of the auditorium position according to the reference sound pressure level and the additional sound pressure level;

assigning the sound pressure level difference value of the auditorium position as a signal gain value to a loudspeaker closest to the auditorium position;

the above steps are repeated until all speakers are given signal gain values.

7. The method of claim 6, wherein said calculating additional sound pressure levels generated by each speaker at each position of an auditorium based on the parameter data for said speaker, the coordinate data for each auditorium position, the coordinate data for each speaker position, and the target area constant value comprises:

calculating the distance value from each loudspeaker to each position of the auditorium according to the coordinate data of each auditorium position and the coordinate data of the position of each loudspeaker;

calculating angle values from the loudspeakers to the positions of the auditorium according to the distance values from the loudspeakers to the positions of the auditorium, the coordinate data of the positions of the loudspeakers and the coordinate data of the intersection of the front axial direction of the loudspeakers and the plane of the auditorium;

and calculating additional sound pressure levels generated by the loudspeakers at the positions of the auditorium according to the angle values from the loudspeakers to the positions of the auditorium, the parameter data of the loudspeakers, the target area constant value and the distance values from the loudspeakers to the positions of the auditorium.

8. The method of claim 6, wherein said calculating a reference sound pressure level for each position of an auditorium from parameter data for said speaker and calculating a sound pressure level difference for the auditorium position from said reference sound pressure level and said additional sound pressure level comprises:

calculating reference sound pressure levels of all positions of the auditorium according to the parameter data of the loudspeakers, and calculating sound pressure levels generated by all the loudspeakers at all the positions of the auditorium according to the reference sound pressure levels and the additional sound pressure levels;

calculating sound pressure generated by each loudspeaker at each position of the auditorium according to the sound pressure level generated by each loudspeaker at each position of the auditorium, and calculating total sound pressure level generated by each loudspeaker at each position of the auditorium according to the sound pressure;

calculating to obtain the average value of the sound pressure level of each position of the auditorium according to the total sound pressure level;

and calculating the sound pressure level difference of the positions of the auditorium according to the total sound pressure level of each position of the auditorium generated by each loudspeaker and the average value of the sound pressure levels of each position of the auditorium.

Technical Field

The present invention relates to the field of conference systems, and in particular, to a speaker signal feeding method for a conference system.

Background

Conference rooms are language-oriented acoustic locations used for meetings, academic reports, learning and training, and the like. In acoustic environment design, in addition to the requirement of sufficiently high speech intelligibility, the requirement is also that the sound field distribution is uniform and the sound image is consistent. In practice, since the speaker's voice attenuates with increasing distance, for most conference rooms, a sound amplification system is installed to increase the sound pressure level at the auditorium.

In order to make the conference room have better effect, the electronic conference system engineering design specification (GB50799-2012) puts forward requirements on each part of the conference system, including classification and composition of the conference public address system, functional design and requirements, performance design requirements, and main equipment design requirements. The acoustic characteristic indexes of the sound amplifying system are also specified in the design specifications of sound amplifying systems of halls and stadiums (GB/T28049 and 2011). The two national standards of electronic conference system engineering construction and quality acceptance standard (GB51043-2014) and hall sound reinforcement characteristic measurement method (GB/T4959-2011) stipulate a measurement acceptance method of a conference room sound reinforcement system.

The conference room sound reinforcement system mainly comprises a microphone (also called a microphone), a tuning and signal processing device, a power amplifier and a loudspeaker. The microphone picks up the sound emitted by the sound source (speech, lecture), which is amplified, processed (equalized, divided, limited, etc.) by the sound signal processing device, then power amplified by the power amplifier, and finally played through the speaker. The speaker arrangement may be selected to be centralized, decentralized or a combination of centralized and decentralized depending on the specific conditions of the conference room.

The system debugging is needed after the conference room sound reinforcement system is designed and installed, and the acoustic characteristic index test comprises the maximum sound pressure level, the transmission frequency characteristic, the sound transmission gain and the non-uniformity of a steady sound field.

All relevant national standards have relevant regulations on the composition, acoustic characteristic indexes, system debugging and system measurement and acceptance of the conference room sound reinforcement system so as to ensure that the constructed conference room has better sound reinforcement performance. However, the relevant national standards and various documents do not describe how to design the signals fed to the loudspeakers of a sound amplification system so that the sound amplification system has high speech intelligibility, uniform sound field distribution and consistent sound image.

In practical engineering applications, the signals fed to the various loudspeakers of a conference system are generally determined in two ways:

firstly, the field debugging is carried out by experience. After the system is installed, the system feeds signals of all the loudspeakers by experience, then acoustic indexes are tested in an auditorium, and the signals of all the loudspeakers are corrected according to test results until the signals meet the requirements.

And secondly, simulation is carried out through computer software, and at present, EASE software is mainly used. Firstly, modeling a room, setting interface materials, speaker types and installation positions, then calculating acoustic indexes at auditorium positions, and carrying out corresponding adjustment according to simulation results to enable the acoustic indexes of the system to meet design requirements.

Although the two common methods for determining the signal fed to the speaker in the conference system can achieve better effect and are widely used, there are some problems, which are mainly shown in the following:

the method of field debugging by experience has two main problems: (1) the debugging personnel need to have very rich experience, can rapidly provide a correction scheme according to a field acoustic index test result and immediately implement the correction scheme, and can meet the requirements only by repeatedly adjusting for many times. (2) The whole system can be installed after the system is installed, and in case of great careless omission in early design, the debugging is difficult to make up.

The main problems of the simulation method by EASE software are: (1) a higher expense is required to purchase the software. (2) Different versions of the calculations differ, and EASE software developers have never published prediction and actual measurement comparison cases to illustrate the accuracy of the software calculations. (3) Only the model of the loudspeaker in the EASE software library can be selected for simulation, and other loudspeakers cannot be simulated.

Furthermore, both of the above methods generally do not consider the problem that the sound image orientation perceived by the listener is not consistent with the actual sound source; in order to make the sound image direction perceived by the listener consistent with the actual sound source to obtain better audio effect, accurate loudspeaker signal delay value and gain value need to be input in the conference system, and the position of the loudspeaker in the conference place needs to be reasonably arranged; however, the conventional conference system application does not have a calculation scheme of a relevant delay value and gain value, and does not reasonably arrange speakers, so that the sound image direction perceived by listeners is inconsistent with an actual sound source, and the audio effect experience is poor.

Disclosure of Invention

The invention provides a loudspeaker signal feeding method for a conference system, which respectively calculates a loudspeaker delay value and a loudspeaker gain value through a target area parameter and a loudspeaker parameter, and reasonably arranges the loudspeakers in a target area, so as to solve the technical problems that no related delay value and gain value calculation scheme exists in the application of the traditional conference system, and the loudspeakers are not reasonably arranged, so that the sound image direction perceived by listeners is consistent with the actual sound source, and further the audio effect experience of a user is improved.

In order to solve the above technical problem, an embodiment of the present invention provides a speaker signal feeding method for a conference system, including:

acquiring space parameters of a target area and constructing a three-dimensional coordinate system;

measuring reverberation time values of the auditorium and the speaking seat in the target area, calculating to obtain a constant value of the target area according to the reverberation time values and the spatial parameters of the target area, and calculating to obtain a farthest distance value between a loudspeaker and the auditorium in the target area according to the constant value of the target area;

within the farthest distance value range, arranging a plurality of loudspeakers in the target area according to a rule and enabling the front axial directions of the loudspeakers to face an auditorium;

acquiring coordinate data of a speech seat position, coordinate data of each auditorium position and coordinate data of positions of all loudspeakers in the three-dimensional coordinate system, and calculating a signal delay value of each loudspeaker according to the coordinate data of the speech seat position, the coordinate data of each auditorium position and the coordinate data of the positions of all loudspeakers in combination with a sound propagation speed;

acquiring parameter data of each loudspeaker in a target area;

calculating to obtain a signal gain value corresponding to each loudspeaker according to the parameter data of the loudspeaker, the coordinate data of each auditorium position, the coordinate data of the position of each loudspeaker and the constant value of the target area;

and uploading the signal delay value and the signal gain value to a conference system, and configuring corresponding signal delay and gain parameters of the regularly arranged loudspeakers.

As a preferable scheme, the measuring reverberation time values of the auditorium and the speech seat in the target region, calculating a target region constant value according to the reverberation time value and a spatial parameter of the target region, and calculating a farthest distance value between a speaker and the auditorium in the target region according to the target region constant value includes:

uniformly selecting a plurality of measuring points in the auditorium, measuring to obtain the reverberation time of each measuring point, and calculating to obtain the reverberation time average value according to the reverberation time of each measuring point;

calculating to obtain an average sound absorption coefficient value of the target area according to the reverberation time average value and the space parameter of the target area;

calculating to obtain a target area constant value according to the average sound absorption coefficient value, the reverberation time average value and the space parameter of the target area;

and calculating the farthest distance value between the loudspeaker and the auditorium in the target area according to the constant value of the target area.

Preferably, the arranging the plurality of speakers in the target area according to a rule includes:

arranging the loudspeakers around the auditorium in a ring shape, wherein the horizontal distance between each loudspeaker and the auditorium closest to the loudspeaker is more than 1 meter, and the height from the ground is more than 1.2 meters;

the maximum horizontal distance between every two adjacent loudspeakers is 2 times of the distance between the corresponding two adjacent seats.

The signal delay value of the loudspeaker is calculated according to the coordinate data of the speaking seat position, the coordinate data of each listening seat position and the coordinate data of the position of each loudspeaker in combination with the sound propagation speed, and the method comprises the following steps:

calculating the sound propagation time from each position of the auditorium to the auditorium according to the coordinate data of the auditorium position and the coordinate data of each auditorium position and combining the sound propagation speed;

sequencing the sound propagation time from each position of the auditorium to the speech seat to obtain the auditorium position corresponding to the longest time value and the auditorium position corresponding to the shortest time value;

calculating the sound propagation time from each loudspeaker to the speech place according to the coordinate data of the speech place position and the coordinate data of the position of each loudspeaker and the sound propagation speed;

sequencing the sound propagation time from each loudspeaker to the speech position to obtain the loudspeaker position corresponding to the long time value and the loudspeaker position corresponding to the shortest time value;

calculating the maximum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the longest sound propagation time and the position of the loudspeaker with the longest sound propagation time, the maximum sound propagation time value from the speech seat to the auditorium and the sound propagation speed;

calculating the minimum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the shortest sound propagation time and the position of the loudspeaker with the shortest sound propagation time, the minimum sound propagation time value from the speech seat to the auditorium and the sound propagation speed; sequencing the distance values from the loudspeakers to the speech seats to obtain a maximum loudspeaker distance value and a minimum loudspeaker distance value;

and calculating respective signal delay values corresponding to the rest of the loudspeakers according to the maximum loudspeaker distance value, the minimum loudspeaker distance value, the maximum signal delay value, the minimum signal delay value and the distance value from each loudspeaker to the speech seat.

Preferably, the calculating, according to the coordinate data of the floor position and the coordinate data of each floor position and in combination with a sound propagation speed, a sound propagation time from each position of the floor to the floor includes:

calculating the distance value from each auditorium to the speech seat according to the coordinate data of the speech seat position and the coordinate data of each auditorium position;

and calculating the sound propagation time from each position of the auditorium to the speech seat according to the distance value from each auditorium to the speech seat and the sound propagation speed.

Preferably, the calculating, according to the coordinate data of the speaking place position and the coordinate data of the position of each speaker, the sound propagation time from each speaker to the speaking place in combination with the sound propagation speed includes:

calculating to obtain the distance value from each loudspeaker to the speech place according to the coordinate data of the speech place position and the coordinate data of the position of each loudspeaker;

and calculating the sound propagation time from each loudspeaker to the speech place according to the distance value from each loudspeaker to the speech place and the sound propagation speed.

Preferably, the parameter data of the speaker includes: the characteristic sensitivity level of the loudspeaker and the directivity coefficient of the axial included angle of the loudspeaker.

As a preferred scheme, the calculating, according to the parameter data of the speakers, the coordinate data of the positions of the auditoriums, the coordinate data of the positions of the speakers, and the constant value of the target area, a signal gain value corresponding to each speaker includes:

calculating additional sound pressure levels generated by the loudspeakers at the positions of the auditoriums according to the parameter data of the loudspeakers, the coordinate data of the positions of the auditoriums, the coordinate data of the positions of the loudspeakers and the constant value of the target area;

calculating the reference sound pressure level of each position of the auditorium according to the parameter data of the loudspeaker, and calculating the sound pressure level difference value of the auditorium position according to the reference sound pressure level and the additional sound pressure level;

assigning the sound pressure level difference value of the auditorium position as a signal gain value to a loudspeaker closest to the auditorium position;

the above steps are repeated until all speakers are given signal gain values.

Preferably, the calculating, according to the parameter data of the speakers, the coordinate data of the positions of the auditoriums, the coordinate data of the positions of the speakers, and the target area constant value, additional sound pressure levels generated by the speakers at the positions of the auditoriums includes:

calculating the distance value from each loudspeaker to each position of the auditorium according to the coordinate data of each auditorium position and the coordinate data of the position of each loudspeaker;

calculating angle values from the loudspeakers to the positions of the auditorium according to the distance values from the loudspeakers to the positions of the auditorium, the coordinate data of the positions of the loudspeakers and the coordinate data of the intersection of the front axial direction of the loudspeakers and the plane of the auditorium;

and calculating additional sound pressure levels generated by the loudspeakers at the positions of the auditorium according to the angle values from the loudspeakers to the positions of the auditorium, the parameter data of the loudspeakers, the target area constant value and the distance values from the loudspeakers to the positions of the auditorium.

Preferably, the calculating a reference sound pressure level of each position of the auditorium according to the parameter data of the loudspeaker, and calculating a sound pressure level difference value of the auditorium position according to the reference sound pressure level and the additional sound pressure level includes:

calculating reference sound pressure levels of all positions of the auditorium according to the parameter data of the loudspeakers, and calculating sound pressure levels generated by all the loudspeakers at all the positions of the auditorium according to the reference sound pressure levels and the additional sound pressure levels;

calculating sound pressure generated by each loudspeaker at each position of the auditorium according to the sound pressure level generated by each loudspeaker at each position of the auditorium, and calculating total sound pressure level generated by each loudspeaker at each position of the auditorium according to the sound pressure;

calculating to obtain the average value of the sound pressure level of each position of the auditorium according to the total sound pressure level;

and calculating the sound pressure level difference of the positions of the auditorium according to the total sound pressure level of each position of the auditorium generated by each loudspeaker and the average value of the sound pressure levels of each position of the auditorium.

Compared with the prior art, the embodiment of the invention has the following beneficial effects:

according to the invention, the delay value and the gain value of the loudspeaker are respectively calculated through the target area parameters and the loudspeaker parameters, and the loudspeaker is reasonably arranged in the target area, so that the technical problems that no related delay value and gain value calculation scheme exists in the application of the traditional conference system and the loudspeaker is not reasonably arranged are solved, and therefore, the sound image direction perceived by a listener is consistent with the actual sound source, and the audio effect experience of a user is improved.

Drawings

FIG. 1: a flow chart of method steps for speaker signal feed in an embodiment of the present invention;

FIG. 2: constructing a schematic diagram of a three-dimensional coordinate system of a target area in the embodiment of the invention;

FIG. 3: the loudspeaker orientation arrangement structure in the embodiment of the invention is a schematic diagram.

Detailed Description

The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.

Referring to fig. 1, a preferred embodiment of the present invention provides a speaker signal feeding method for a conference system, including:

s1, acquiring the space parameters of the target area and constructing a three-dimensional coordinate system;

measuring the space size of a room with the room as a target area, wherein the space size is recorded as L, W, H and the unit is meter; an O-XYZ coordinate system is established on a room, an O-XY plane is the ground, an X axis is a long side, a Y axis is a short side, and a Z axis is in a vertical direction. As shown in fig. 2.

S2, measuring reverberation time values of the auditorium and the speech seat in the target area, calculating to obtain a target area constant value according to the reverberation time values and the space parameters of the target area, and calculating to obtain a farthest distance value between a loudspeaker and the auditorium in the target area according to the target area constant value;

in this embodiment, the step S2 specifically includes:

s21, uniformly selecting a plurality of measuring points in the auditorium, measuring to obtain the reverberation time of each measuring point, and calculating to obtain the reverberation time average value according to the reverberation time of each measuring point;

the reverberation time of the auditorium and the speaking seat in the room is measured, 3-6 measuring points of the auditorium can be uniformly taken, and the height of the test microphone is about 1.2 meters. The reverberation time of each measuring point is recorded as RT_i(i is 1,2, … …, n.n. is the number of stations), and the average value of the reverberation time is denoted as RT.

S22, calculating an average sound absorption coefficient value of the target area according to the reverberation time average value and the space parameter of the target area;

calculate room average sound absorption coefficient α:

s23, calculating to obtain a target area constant value according to the average sound absorption coefficient value, the reverberation time average value and the space parameter of the target area;

calculating the room constant R:

and S24, calculating the farthest distance value between the loudspeaker and the auditorium in the target area according to the constant value of the target area.

Calculating the farthest distance r between the loudspeaker and the auditorium (including the auditorium and the speech seat)_max：

Wherein β is coefficient with value range of 0.15-0.5 and default value of 0.4, and the distance between each speaker and the listener's seat farthest from each speaker is less than r_max。

S3, arranging a plurality of speakers in the target area according to a rule and making the front axes of the speakers face the auditorium in the farthest distance value range, as shown in fig. 3; the coordinate of the front axis of each speaker (identified by reference numeral j) intersecting the plane of the listener's seat is denoted by (x)_j0，y_j0,z_j0)

In this embodiment, the regularly arranging the plurality of speakers in the target area includes:

s31, arranging the loudspeakers around the auditorium in a ring shape, wherein the horizontal distance between each loudspeaker and the auditorium closest to the loudspeaker is more than 1 m, and the height from the ground is more than 1.2 m;

and S32, horizontally separating every two adjacent loudspeakers by a maximum distance which is 2 times of the distance between the corresponding two adjacent seats.

In addition, some speakers may be arranged on the ceiling.

S4, acquiring coordinate data of a speech seat position, coordinate data of each auditorium position and coordinate data of positions of all loudspeakers in the three-dimensional coordinate system, and calculating a signal delay value of each loudspeaker according to the coordinate data of the speech seat position, the coordinate data of each auditorium position and the coordinate data of the positions of all loudspeakers in combination with sound propagation speed;

first, coordinates (x) of a floor position are acquired_s，y_s,z_s) Coordinates (x) of each auditorium position_i，y_i,z_i) Coordinates (x) of the location of each loudspeaker_j，y_j,z_j) Where i is 1,2, … M, j is 1,2, … P, M, P are the number of auditoriums and loudspeakers, respectively. Wherein z is_s，z_i,z_jIs equal, taking a value close to the human sitting height, for example 1.2 meters.

Then, in this embodiment, the calculating a signal delay value of the speaker according to the coordinate data of the floor position, the coordinate data of each audience position, and the coordinate data of the position where each speaker is located, in combination with the sound propagation speed specifically includes:

the signal delay value of the loudspeaker is calculated according to the coordinate data of the speaking seat position, the coordinate data of each listening seat position and the coordinate data of the position of each loudspeaker in combination with the sound propagation speed, and the method comprises the following steps:

calculating the sound propagation time from each position of the auditorium to the auditorium according to the coordinate data of the auditorium position and the coordinate data of each auditorium position and combining the sound propagation speed;

sequencing the sound propagation time from each position of the auditorium to the speech seat to obtain the auditorium position corresponding to the longest time value and the auditorium position corresponding to the shortest time value;

calculating the sound propagation time from each loudspeaker to the speech place according to the coordinate data of the speech place position and the coordinate data of the position of each loudspeaker and the sound propagation speed;

sequencing the sound propagation time from each loudspeaker to the speech position to obtain the loudspeaker position corresponding to the longest time value and the loudspeaker position corresponding to the shortest time value;

calculating the maximum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the longest sound propagation time and the position of the loudspeaker with the longest sound propagation time, the maximum sound propagation time value from the speech seat to the auditorium and the sound propagation speed;

calculating the minimum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the shortest sound propagation time and the position of the loudspeaker with the shortest sound propagation time, the minimum sound propagation time value from the speech seat to the auditorium and the sound propagation speed; sequencing the distance values from the loudspeakers to the speech seats to obtain a maximum loudspeaker distance value and a minimum loudspeaker distance value;

and calculating respective signal delay values corresponding to the rest of the loudspeakers according to the maximum loudspeaker distance value, the minimum loudspeaker distance value, the maximum signal delay value, the minimum signal delay value and the distance value from each loudspeaker to the speech seat.

S41, calculating the sound propagation time from each position of the auditorium to the speech seat according to the coordinate data of the speech seat position and the coordinate data of each auditorium position and combining the sound propagation speed;

the method specifically comprises the following steps:

s411, calculating the distance value from each auditorium to the speech seat according to the coordinate data of the speech seat position and the coordinate data of each auditorium position; calculate the distance of each auditorium to the floor:

and S412, calculating the sound propagation time from each position of each auditorium to the speech seat according to the distance value from each auditorium to the speech seat and the sound propagation speed. Calculating the sound propagation time from each position of the auditorium to the speech position:

the speed of sound in air is about 340m/s, which is 340 in this embodiment.

S42, sequencing the sound propagation time from each position of the auditorium to the speech seat to obtain the auditorium position corresponding to the longest time value and the auditorium position corresponding to the shortest time value;

for t_iSorting, finding the maximum and minimum values max (t)_i)、min(t_i) The corresponding auditorium position is denoted A, B.

S43, calculating the sound propagation time from each loudspeaker to the speech place according to the coordinate data of the speech place position and the coordinate data of the position of each loudspeaker and the sound propagation speed;

the method specifically comprises the following steps:

s431, calculating the distance value from each loudspeaker to the speech place according to the coordinate data of the speech place position and the coordinate data of the position of each loudspeaker; calculating the distance from each loudspeaker to the speech floor:

and S432, calculating the sound propagation time from each loudspeaker to the speech seat according to the distance value from each loudspeaker to the speech seat and the sound propagation speed. Calculating the sound propagation time between each loudspeaker and the speech place:

the speed of sound in air is about 340m/s, which is taken as 340 in this example.

S44, sequencing the sound propagation time from each loudspeaker to the speech seat to obtain the loudspeaker position corresponding to the longest time value and the loudspeaker position corresponding to the shortest time value;

for t_jSorting, finding the maximum and minimum values max (t)_j)、min(t_j) The corresponding speaker location is labeled C, D.

S45, calculating the maximum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the longest sound propagation time and the position of the loudspeaker with the longest sound propagation time, the maximum sound propagation time value from the speech seat to the auditorium and the sound propagation speed; calculating the feed max (t)_j) Corresponding signal delay value T of loudspeaker_max：

Wherein AC refers to the distance between the positions A, C, tau is an additional value, the value range is 0.01-0.02, and the default value is 0.015.

S46, calculating the minimum signal delay value of the corresponding loudspeaker according to the distance value between the position of the auditorium with the shortest sound propagation time and the position of the loudspeaker with the shortest sound propagation time, the minimum sound propagation time value from the speech seat to the auditorium and the sound propagation speed; calculating feed min (t)_j) Corresponding signal delay value T of loudspeaker_min：

The BD refers to the distance between the positions B, D, τ is an additional value, the value range is 0.01-0.02, and the default value is 0.015.

S47, sorting the distance values from the speakers to the speech seats to obtain a maximum distance value of the speakers and a minimum distance value of the speakers;

to r_jSorting to find the maximum and minimum values max (r)_j)、min(r_j)。

S48, according to the maximum distance, the minimum distance and the maximum signal delay of the loudspeakerAnd calculating the values, the minimum signal delay values and the distance values from all the loudspeakers to the speech place to obtain the respective signal delay values corresponding to the rest loudspeakers. Calculating the value of the signal delay T to feed the remaining loudspeakers_j：

S5, acquiring parameter data of each loudspeaker in the target area;

in this embodiment, the parameter data of the speaker includes:

loudspeaker characteristic sensitivity level L_EIn dB;

and the directivity coefficient D (theta) forms an angle theta with the axial direction of the loudspeaker.

S6, calculating to obtain signal gain values corresponding to the loudspeakers according to the parameter data of the loudspeakers, the coordinate data of the positions of the auditoriums, the coordinate data of the positions of the loudspeakers and the constant values of the target area;

in this embodiment, the step S6 specifically includes:

s61, calculating additional sound pressure levels generated by the loudspeakers at the positions of the auditoriums according to the parameter data of the loudspeakers, the coordinate data of the positions of the auditoriums, the coordinate data of the positions of the loudspeakers and the constant value of the target area;

in this embodiment, the step S61 specifically includes:

s611, calculating to obtain distance values from each loudspeaker to each position of the auditorium according to the coordinate data of each auditorium position and the coordinate data of each loudspeaker position; calculating the distance r from each loudspeaker j to the respective position i of the auditoriumⁱ _j：

S612, according to the distance value from each loudspeaker to each position of the auditorium, the coordinate data of each auditorium position and the coordinate data of each loudspeaker positionCalculating the angle value from each loudspeaker to each position of the auditorium according to the coordinate data of the intersection of the front axial direction of each loudspeaker and the plane of the auditorium; calculating the angle theta of each loudspeaker j to the position i of the auditorium_ij。

Calculating the coordinate (x) of the front axial direction of each loudspeaker j intersected with the plane of the auditorium_j0，y_j0,z_j0) Distances r to the loudspeakers j and to the individual positions i of the auditorium_jj0、r_ij0。

And S613, calculating additional sound pressure levels generated by the loudspeakers at the positions of the auditorium according to the angle values from the loudspeakers to the positions of the auditorium, the parameter data of the loudspeakers, the target area constant values and the distance values from the loudspeakers to the positions of the auditorium.

The additional sound pressure level produced by each loudspeaker j at the respective position i of the auditorium is calculated:

wherein G is_jThe initial value of the preset signal gain value (relative to the original speaker signal received by the microphone) for each speaker is set to 0 dB.

S62, calculating the reference sound pressure level of each position of the auditorium according to the parameter data of the loudspeaker, and calculating the sound pressure level difference of the auditorium position according to the reference sound pressure level and the additional sound pressure level;

in this embodiment, the step S62 specifically includes:

s621, calculating reference sound pressure levels of all positions of the auditorium according to the parameter data of the loudspeakers, and calculating sound pressure levels generated by all the loudspeakers at all the positions of the auditorium according to the reference sound pressure levels and the additional sound pressure levels;

calculating a reference sound pressure level for each position of the auditorium:

L_P0＝L_E+L₀

wherein L is_EIs the characteristic sensitivity level of the loudspeaker, L₀For the correction value, any value can be set, and the default value is 10 dB.

The sound pressure level produced by each loudspeaker j at each position i of the auditorium is calculated:

L_ij＝L_P0+ΔL_ij

s622, calculating sound pressure generated by each loudspeaker at each position of the auditorium according to the sound pressure level generated by each loudspeaker at each position of the auditorium, and calculating total sound pressure level generated by each loudspeaker at each position of the auditorium according to the sound pressure;

the sound pressure generated by each loudspeaker j at each position i of the auditorium is calculated:

calculating the total sound pressure generated by each position i of the auditorium:

the total sound pressure level generated by each position i of the auditorium is calculated:

L_i＝20lgp_i+94；

wherein, the calculation process of the parameter 94 is as follows: according to the sound pressure level calculation formula:

s623, calculating to obtain the average value of the sound pressure level of each position of the auditorium according to the total sound pressure level;

calculating the average value of sound pressure levels of all positions of the auditorium:

and S624, calculating the sound pressure level difference value of the positions of the auditorium according to the total sound pressure level generated by each loudspeaker at each position of the auditorium and the average value of the sound pressure levels of each position of the auditorium.

Calculating the difference between the sound pressure level of each position i of the auditorium and the average value:

s63, assigning the sound pressure level difference value of the auditorium position as a signal gain value to the loudspeaker closest to the auditorium position;

modifying the signal gain value G for each loudspeaker_jΔ L of the audience position i to be closest to the speaker j_iIs given to G_j。

S64, repeating the above steps until all speakers are given signal gain values.

And S7, uploading the signal delay value and the signal gain value to a conference system, and configuring corresponding signal delay and gain parameters for the regularly arranged loudspeakers.

The invention designs a loudspeaker signal feeding method aiming at the sound amplification application scene of a conference system, so that the sound pressure distribution of the auditorium is uniform, and the auditorium perceives that the sound comes from an actual sound source. The technical problems to be solved include:

1. and calculating the signal gain value of each loudspeaker according to the meeting room interface characteristics and the loudspeaker arrangement.

2. The signal delay values of the individual loudspeakers are calculated on the basis of the loudspeaker arrangement.

3. The sound pressure level at the auditorium is calculated.

The technical scheme of the invention has the beneficial effects that:

1. the signal delay and gain of each loudspeaker can be calculated only by the size of the room where the conference system is located, the reverberation time parameter, the sensitivity level of the loudspeaker and the directivity coefficient.

2. The signal delay and gain of each speaker can be calculated so that the sound level heard by each position of the auditorium is basically consistent, and the sound direction comes from the speaker.

3. The system can be designed into computer software or a mobile phone APP, the room size, reverberation time parameters, the sensitivity level of the loudspeaker and the directivity coefficient are input through an interface, the signal delay and the gain of each loudspeaker can be immediately calculated, and the signal delay and the gain are transmitted to a conference system for automatic setting, so that the debugging efficiency of the conference system is greatly improved, and a better sound effect is obtained.

The technical scheme has the advantages that:

1. the sound field of the auditorium is uniformly distributed, and the sound direction perceived by the auditorium is consistent with the actual sound source;

2. the loudspeaker signal feed of the conference system becomes simpler and more convenient, and the signal simulation does not need to be carried out by experienced professionals and purchasing software and hardware equipment with extra expense.

The above-mentioned embodiments are provided to further explain the objects, technical solutions and advantages of the present invention in detail, and it should be understood that the above-mentioned embodiments are only examples of the present invention and are not intended to limit the scope of the present invention. It should be understood that any modifications, equivalents, improvements and the like, which come within the spirit and principle of the invention, may occur to those skilled in the art and are intended to be included within the scope of the invention.