CN111711914A - Sound amplification system with reverberation time measuring function - Google Patents
Sound amplification system with reverberation time measuring function Download PDFInfo
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- CN111711914A CN111711914A CN202010544933.2A CN202010544933A CN111711914A CN 111711914 A CN111711914 A CN 111711914A CN 202010544933 A CN202010544933 A CN 202010544933A CN 111711914 A CN111711914 A CN 111711914A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R27/00—Public address systems
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- G—PHYSICS
- G01—MEASURING; TESTING
- G01H—MEASUREMENT OF MECHANICAL VIBRATIONS OR ULTRASONIC, SONIC OR INFRASONIC WAVES
- G01H7/00—Measuring reverberation time ; room acoustic measurements
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2227/00—Details of public address [PA] systems covered by H04R27/00 but not provided for in any of its subgroups
Abstract
The invention discloses a sound amplification system with a reverberation time measuring function, which comprises a sound amplification sound box, a microphone and an audio host; the audio host measures reverberation time by adopting an exponential sweep signal as an impulse response integration method of an excitation sound source; the microphone is used for transmitting the collected audio signals to the audio host; the audio host performs linear convolution on the acquired audio signal and the reverse frequency sweeping signal to obtain an impulse response signal; then, the impulse response signals pass through octave filters with different central frequencies respectively to obtain corresponding impulse response signals under different central frequencies; and finally, performing inverse integration on all the impulse response signals to obtain an energy attenuation curve, selecting sample data in the energy attenuation curve to perform linear fitting, and calculating reverberation time according to the slope of a linear fitting straight line. The invention has the advantages of strong practicability, high precision, small error, short measurement time, capability of simultaneously measuring a plurality of different positions, capability of remote measurement and the like.
Description
Technical Field
The invention relates to the technical field of sound amplification systems, in particular to a sound amplification system with a reverberation time measuring function.
Background
The public address system generally refers to a system for amplifying the voice of a speaker to a talker in real time, so that the speaker and the talker are in the same acoustic environment, and the public address system includes a public address system and a sound field. A successful sound amplification system must have sufficient loudness and sufficient intelligibility and be able to cover the audience area evenly with sound. The sound reinforcement system is widely applied to places such as classrooms, meeting rooms, gymnasiums, cinemas and the like.
Studies have shown that reverberation time is one of the acoustic parameters that affect the sound field and intelligibility in the room. A proper reverberation time contributes to the homogeneity and intelligibility of the sound field, whereas if the reverberation time is too large the intelligibility will be degraded, resulting in a poor listening experience. When the sound amplifying system is actually used, the on-site sound amplifying definition is often reduced due to overlarge reverberation time, but the current measurement of the indoor reverberation time needs a professional system and a professional person, so that most application occasions cannot accurately judge the influence of the indoor reverberation time on the sound quality.
Different types of buildings specified in national standard GB50118-2010 in China need to meet different reverberation time requirements during design, for example, in the design of buildings in schools, the volume is 200m3In the following ordinary classrooms, the air field reverberation time (500 Hz-1 kHz) should not be more than 0.8 s. For example, in the design of office buildings, the volume is 200m3In the following video conference room, the empty field reverberation time (500 Hz-1 kHz) should not be more than 0.6 s.
Reverberation time measurement is generally carried out by two methods, namely, an interrupted sound source method and an impulse response integration method. The frequency range of the measurement depends on the purpose of the measurement, the frequency range of the simple level measurement is at least 250Hz to 2000Hz, and the frequency range of the engineering level and the precision level measurement is 125Hz to 4000 Hz. The sound source interruption method is a method of directly recording a sound pressure level attenuation curve by suddenly cutting off a sound source to stop sounding after the sounding of the sound source reaches a stable state. The greatest disadvantage of this method is that the acoustic decay is influenced by the transient fluctuations that are unavoidable in the stochastic process. The impulse response integral rule can well solve the instantaneous fluctuation problem in the random process, the obtained attenuation curve is smooth and has small fluctuation, and the measurement result is more accurate. Therefore, it is more practical and convenient to measure the reverberation time using the impulse response integration method than the interrupted sound source method.
In the impulse response integration method, an impulse response from a sound source position to a receiver position is generally obtained by two methods, one is a direct method such as directly obtaining an impulse response using a pistol, a pulsed electric spark, a balloon, or the like. The other method is indirect method, and the impulse response is indirectly obtained through a frequency sweep signal or an MLS signal. The frequency sweep signal has many advantages as an excitation signal for reverberation time measurement, for example, the sensitivity to system time variation (temperature variation and air flow) can be reduced, and the reduction of the effective signal-to-noise ratio due to harmonic distortion can be avoided. The feed power of the sinusoidal swept excitation signal is actually larger than the MLS signal, since all harmonic distortion can be filtered out of the result. In a quiet environment, the sweep measurement may provide a signal-to-noise ratio of greater than 100 dB. Therefore, the method for measuring the reverberation time by adopting the exponential sweep frequency signal as the impulse response integration method of the excitation sound source has the advantages of small error, high repeatability, high signal-to-noise ratio and the like.
Disclosure of Invention
The invention aims to provide an acoustic amplification system with a reverberation time measuring function aiming at the defects of the prior art.
The purpose of the invention is realized by the following technical scheme: a sound amplification system with a reverberation time measuring function comprises a sound amplification sound box, a microphone, an audio host and the like; the audio host computer measures reverberation time by adopting an exponential sweep signal as an impulse response integration method of an excitation sound source.
The sound amplification sound box is used for playing an index frequency sweep signal;
the microphone is used for receiving audio signals and transmitting the received audio signals to the audio host;
the audio host comprises an analog-to-digital conversion module (A/D), a digital processing unit (DSP) and a digital-to-analog conversion module (D/A); the analog-to-digital conversion module (A/D) converts the received audio signal into a digital signal which can be directly processed by a digital signal processing unit (DSP); the digital signal processing unit (DSP) processes the transmitted digital signal, and the specific processing steps are as follows:
(1) generating an exponential sweep signal, wherein the generation formula is as follows:
wherein x (t) represents an exponential sweep signal, f1Denotes the starting frequency, f2Denotes the cut-off frequency, T denotes the duration of the signal, ln (-) denotes the natural logarithm, exp (-) denotes the natural exponent, T denotes the moment of the sampling point correspondence.
(2) Generating a reverse frequency sweeping signal, wherein the generation formula is as follows:
e(t)＝fliplr(x(t))
wherein e (t) represents the reverse sweep frequency signal, and the fliplr (·) represents the flipping operation, i.e. the position of the first sampling point of x (t) is exchanged with the last sampling point, the position of the second sampling point is exchanged with the second last sampling point, and so on.
(3) Using the sound box playing index frequency sweep signal x (t), carrying out linear convolution on the audio signals received by the plurality of microphones and the reverse frequency sweep signal e (t) respectively to obtain an impulse response signal, wherein the calculation formula is as follows:
wherein h isi(t) represents the impulse response signal of the ith microphone, yi(t) represents the audio signal received by the ith microphone, i is the serial number of the microphone,
representing a linear convolution operation.
(4) And respectively passing the impulse response signals through octave filters with different central frequencies to obtain corresponding impulse response signals under different central frequencies.
(5) And performing reverse integration on the impulse response signals corresponding to different central frequencies to obtain an energy attenuation curve, wherein the calculation formula is as follows:
wherein h isi,f(τ) represents an impulse response signal corresponding to the center frequency f of the ith microphone, Ei,f(t) represents the energy of inverse integration of the impulse response signal corresponding to the center frequency f of the ith microphone,
representing an integration operation.
(6) Will Ei,f(t) is converted to dB, and the calculation formula is as follows:
Ei,f_dB＝10*log10(Ei,f(t))
(7) selection of Ei,fAnd performing linear fitting on sample data between-5 dB and-25 dB in _dB, wherein the linear equation expression after linear fitting is as follows:
yyf＝kf*xf+bf
wherein, yyfDependent variable, x, representing a linear equationfIndependent variable, k, representing the equation of a straight linefRepresenting the slope of the linear equation, bfRepresenting the intercept of the straight-line equation.
(8) Calculating the reverberation time corresponding to the center frequency f, wherein the calculation formula is as follows:
wherein, Ti,fDenotes the reverberation time, k, corresponding to the center frequency f of the ith microphonei,fIndicating the slope of the fitted line corresponding to the center frequency f of the ith microphone.
Furthermore, the measurement signal of the system is an exponential frequency sweep signal, the frequency range is 20 Hz-16 kHz, and the duration is 8 s.
Further, the center frequencies measured by the system are 125Hz, 250Hz, 500Hz, 1000Hz, 2000Hz, and 4000 Hz.
Further, the center frequency of the octave filter is 125Hz, 250Hz, 500Hz, 1000Hz, 2000Hz, 4000 Hz.
Furthermore, the sound amplifying system also comprises a computer which is used for sending the test instruction and displaying the measurement result of the reverberation time.
Furthermore, the audio host also comprises a power amplifier, so that the output of good tone quality is effectively ensured.
Further, the audio host may connect 8 microphones for measuring reverberation times of 8 locations within the room at the same time.
The invention has the beneficial effects that: the invention transplants the function of measuring the reverberation time into the sound amplifying system, so that the sound amplifying system has the function of measuring the reverberation time, and the frequency range of the measured reverberation time accords with the characteristics of engineering level and precision level. The invention can be directly used for measuring the reverberation time of the closed hall. The index frequency sweep signal used by the invention has the advantages of repeatability, high signal-to-noise ratio, strong anti-interference capability, high accuracy and the like. In addition, the invention can simultaneously measure the reverberation time of 8 different positions, thereby effectively reducing the time for measuring the reverberation time. The invention also supports the remote reverberation time measuring function, namely, the reverberation time is measured by a network remote operating system, and the result is returned to the test page. In a word, the reverberation time measuring function in the invention has the advantages of strong practicability, high precision, small error, short measuring time, remote measurement, simultaneous measurement of 8 different positions and the like.
Drawings
Fig. 1 is a hardware configuration diagram of the present invention.
Fig. 2 is a diagram of the reverberation time calculation process of the present invention.
FIG. 3 is a view showing the arrangement of the apparatus according to the present invention during measurement.
Detailed Description
The following describes embodiments of the present invention in further detail with reference to the accompanying drawings.
As shown in fig. 1, a sound amplification system with reverberation time measurement function mainly comprises five parts, which are a microphone, a sound box, an audio host, a switch, a computer, and the like. The audio host is divided into modules such as analog-to-digital conversion (A/D), a digital signal processing unit (DSP), digital-to-analog conversion (D/A), a power amplifier and the like.
The method comprises the following specific implementation steps:
step 1: under the remote condition, a remote computer is connected to an audio host of a room to be tested through a switch and a network, a test command is issued to the audio host through the computer, and the audio host can execute the following operation according to the test command.
Step 2: generating an exponential sweep signal, wherein the generation formula is as follows:
wherein x (t) represents an exponential sweep signal, f1Denotes the starting frequency, f2Denotes the cut-off frequency, T denotes the duration of the signal, ln (-) denotes the natural logarithm, exp (-) denotes the natural exponent, T denotes the moment of the sampling point correspondence.
And step 3: generating a reverse frequency sweeping signal, wherein the generation formula is as follows:
e(t)＝fliplr(x(t))
wherein e (t) represents the reverse sweep frequency signal, and the fliplr (·) represents the flipping operation, i.e. the position of the first sampling point of x (t) is exchanged with the last sampling point, the position of the second sampling point is exchanged with the second last sampling point, and so on.
And 4, step 4: using the sound box playing index frequency sweep signal x (t), respectively carrying out linear convolution on the audio signals received by the 8 microphones and the reverse frequency sweep signal e (t) to obtain an impulse response signal, wherein the calculation formula is as follows:
wherein h isi(t) represents the impulse response signal of the ith microphone, yi(t) represents the audio signal received by the ith microphone, i is the serial number of the microphone and takes the value of 1-8,
representing a linear convolution operation.
And 5: and respectively passing the impulse response signals through octave filters with different central frequencies to obtain corresponding impulse response signals under different central frequencies.
Step 6: and performing reverse integration on the impulse response signal corresponding to the center frequency f to obtain an energy attenuation curve, wherein the calculation formula is as follows:
wherein h isi,f(τ) represents an impulse response signal corresponding to the center frequency f of the ith microphone, Ei,f(t) represents the energy of inverse integration of the impulse response signal corresponding to the center frequency f of the ith microphone,
representing an integration operation.
And 7: decay the energy curve Ei,f(t) is converted to dB, and the calculation formula is as follows:
Ei,f_dB＝10*log10(Ei,f(t))
and 8: selecting an energy attenuation curve Ei,fAnd performing linear fitting on sample data between-5 dB and-25 dB in _dB, wherein the linear equation expression after linear fitting is as follows:
yyf＝kf*xf+bf
wherein, yyfDependent variable, x, representing a linear equationfIndependent variable, k, representing the equation of a straight linefRepresenting the slope of the linear equation, bfRepresenting the intercept of the straight-line equation.
And step 9: calculating the reverberation time corresponding to the center frequency f, wherein the calculation formula is as follows:
wherein, Ti,fDenotes the reverberation time, k, corresponding to the center frequency f of the ith microphonei,fIndicating the slope of the fitted line corresponding to the center frequency f of the ith microphone.
Step 10: the reverberation time values corresponding to different frequencies obtained by the calculation of the steps are transmitted to a remote computer through a network.
Measurement example: the reverberation time in a certain room is measured.
The sound box and 8 microphones included in the invention are placed in a room to be tested, the placement positions are shown in figure 3, a tester is remotely connected with an audio host computer in the room to be tested through a network to control, the generated exponential frequency sweep signal is played through the sound box, the microphones are adopted at the 8 positions to be tested to receive audio signals responded by the room, a DSP module in the audio host computer performs linear convolution, filtering, inverse integration and linear fitting on the signals to calculate 8 × 6 reverberation times Ti,fThe measurement results are shown in table 1.
TABLE 1 reverberation time measured by the invention
The invention measures the reverberation time by generating the index sweep frequency signal, not only can remotely measure, but also can simultaneously measure the reverberation time at 8 different positions. The invention can simultaneously measure the reverberation time of 8 different positions in each measurement, thereby saving the measurement time. When the device is used for remote measurement, the influence of personnel on the test environment is reduced, so that the error of the measurement result is smaller.
The above-described embodiments are intended to illustrate rather than to limit the invention, and any modifications and variations of the present invention are within the spirit of the invention and the scope of the appended claims.
Claims (7)
1. A sound amplification system with a reverberation time measuring function is characterized by comprising a sound amplification sound box, a microphone, an audio host and the like; the audio host computer measures reverberation time by adopting an exponential sweep signal as an impulse response integration method of an excitation sound source.
The sound amplification sound box is used for playing an index frequency sweep signal;
the microphone is used for receiving audio signals and transmitting the received audio signals to the audio host;
the audio host comprises an analog-to-digital conversion module (A/D), a digital processing unit (DSP) and a digital-to-analog conversion module (D/A); the analog-to-digital conversion module (A/D) converts the received audio signal into a digital signal which can be directly processed by a digital signal processing unit (DSP); the digital signal processing unit (DSP) processes the transmitted digital signal, and the specific processing steps are as follows:
(1) generating an exponential sweep signal, wherein the generation formula is as follows:
wherein x (t) represents an exponential sweep signal, f1Denotes the starting frequency, f2Denotes the cut-off frequency, T denotes the duration of the signal, ln (-) denotes the natural logarithm, exp (-) denotes the natural exponent, T denotes the moment of the sampling point correspondence.
(2) Generating a reverse frequency sweeping signal, wherein the generation formula is as follows:
e(t)＝fliplr(x(t))
wherein e (t) represents the reverse sweep frequency signal, and the fliplr (·) represents the flipping operation, i.e. the position of the first sampling point of x (t) is exchanged with the last sampling point, the position of the second sampling point is exchanged with the second last sampling point, and so on.
(3) Using the sound box playing index frequency sweep signal x (t), carrying out linear convolution on the audio signals received by the plurality of microphones and the reverse frequency sweep signal e (t) respectively to obtain an impulse response signal, wherein the calculation formula is as follows:
wherein h isi(t) represents the impulse response signal of the ith microphone, yi(t) represents the audio signal received by the ith microphone, i is the serial number of the microphone,
representing a linear convolution operation.
(4) And respectively passing the impulse response signals through octave filters with different central frequencies to obtain corresponding impulse response signals under different central frequencies.
(5) And performing reverse integration on the impulse response signals corresponding to different central frequencies to obtain an energy attenuation curve, wherein the calculation formula is as follows:
wherein h isi,f(τ) represents an impulse response signal corresponding to the center frequency f of the ith microphone, Ei,f(t) represents the energy of inverse integration of the impulse response signal corresponding to the center frequency f of the ith microphone,
representing an integration operation.
(6) Will Ei,f(t) is converted to dB, and the calculation formula is as follows:
Ei,f_dB＝10*log10(Ei,f(t))
(7) selection of Ei,fAnd performing linear fitting on sample data between-5 dB and-25 dB in _dB, wherein the linear equation expression after linear fitting is as follows:
yyf＝kf*xf+bf
wherein, yyfDependent variable, x, representing a linear equationfIndependent variable, k, representing the equation of a straight linefRepresenting the slope of the linear equation, bfRepresenting the intercept of the straight-line equation.
(8) Calculating the reverberation time corresponding to the center frequency f, wherein the calculation formula is as follows:
wherein, Ti,fDenotes the reverberation time, k, corresponding to the center frequency f of the ith microphonei,fIndicating the slope of the fitted line corresponding to the center frequency f of the ith microphone.
2. The system of claim 1, wherein the measurement signal of the system is an exponential frequency sweep signal, the frequency range is 20Hz to 16kHz, and the duration is 8 s.
3. The system of claim 1, wherein the center frequency of the system measurement is 125Hz, 250Hz, 500Hz, 1000Hz, 2000Hz, 4000 Hz.
4. The system of claim 1, wherein the octave filter has a center frequency of 125Hz, 250Hz, 500Hz, 1000Hz, 2000Hz, 4000 Hz.
5. The sound amplifying system with the function of measuring the reverberation time as claimed in claim 1, wherein the sound amplifying system further comprises a computer for issuing a test command and displaying the measurement result of the reverberation time.
6. The system of claim 1, wherein the audio host further comprises a power amplifier, which effectively ensures good sound quality output.
7. The system of claim 1, wherein the audio host is connectable to 8 microphones for simultaneously measuring reverberation time of 8 locations in the room.
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