CN109862503A - A kind of method and apparatus of loudspeaker delay adjust automatically - Google Patents
A kind of method and apparatus of loudspeaker delay adjust automatically Download PDFInfo
- Publication number
- CN109862503A CN109862503A CN201910089278.3A CN201910089278A CN109862503A CN 109862503 A CN109862503 A CN 109862503A CN 201910089278 A CN201910089278 A CN 201910089278A CN 109862503 A CN109862503 A CN 109862503A
- Authority
- CN
- China
- Prior art keywords
- loudspeaker
- delay
- audio signal
- measured
- window
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
Abstract
The present invention provides a kind of method and apparatus of loudspeaker delay parameter adjust automatically, wherein the described method includes: the audio signal after the impulse modulation after tested that any loudspeaker to be measured issues in receiving chamber；It is the starting testing time with the time that the loudspeaker issues audio signal, convolution algorithm is carried out to the audio signal and test pulse signal that receive at position receiving, the loudspeaker is obtained and reaches the relative time delay received at position；The corresponding delay parameter of each loudspeaker to be measured is calculated according to the corresponding relative time delay of all loudspeakers in interior to be measured；Delay adjustment is carried out to loudspeaker corresponding to the delay parameter according to the delay parameter.It solves the problems, such as large-scale cabinet speaker delay adjustment through the above scheme, can reach and save delay adjustment human cost and time cost, the technical effect for improving delay Adjustment precision.
Description
Technical field
The present invention relates to field of voice signal, in particular to a kind of method of loudspeaker delay adjust automatically with set
It is standby.
Background technique
In the large-scale indoor scenarios such as movie theatre, meeting room, stage and KTV large size parlor, after newly built construction construction is completed, by
It is larger in indoor scenarios, often there are multiple loudspeakers and place apart from each other and different apart from LisPos, leads to multiple loudspeakings
Device reaches the acoustic informations of LisPos after sounding at the same time, and there are different delays, this can not only generate phase deviation, cause
The strength reduction of acoustic information, and due to the superposition of different delayed time, it will lead to the clarity decline of acoustic information, greatly drop
Low hearing effect and user experience.
Under normal circumstances, engineering staff can utilize the methods of eye estimate, range measurement, by virtue of experience to large-scale indoor
Different delay parameters is arranged in each loudspeaker, so that the acoustic information for issuing indoor each loudspeaker reaches LisPos
Delay is corresponding.For example, some engineering staffs can test distance of each loudspeaker with respect to LisPos using infrared range-measurement system, then
The delay that each loudspeaker is issued to from acoustic information LisPos is calculated by the speed formula of sound transmission, is further counted
Each loudspeaker is calculated to need while reaching the delay being superimposed needed for LisPos.The method of manual measurement not only accuracy of measurement
It is not high, and engineering staff's voluntarily computation delay is needed, the degree of automation is low, inefficiency, and it is low to exist simultaneously delay accuracy rate
The problem of.
Currently, in view of the above-mentioned problems, proposing a kind of technical solution that can effectively solve the problem that problem not yet.
Summary of the invention
The present invention provides a kind of method and apparatus of loudspeaker delay adjust automatically, realizes that large-scale indoor multiple loudspeakers prolong
When adjust automatically target, reach the adjustment that large-scale indoor speaker delay is realized in automation, improve the accuracy of delay adjustment, save
The time cost of adjustment and the technical effect of human cost.
On the one hand, the present invention provides a kind of methods of loudspeaker delay adjust automatically, comprising:
Audio signal after the impulse modulation after tested that any loudspeaker to be measured issues in receiving chamber；
It is the starting testing time with the time that the loudspeaker issues audio signal, is receiving at position to the institute received
It states audio signal and test pulse signal carries out convolution algorithm, obtain the loudspeaker and reach the relative time delay received at position；
The corresponding delay of each loudspeaker to be measured is calculated according to the corresponding relative time delay of all loudspeakers in interior to be measured
Parameter；
Delay adjustment is carried out to loudspeaker corresponding to the delay parameter according to the delay parameter.
In one embodiment, the method that the audio signal and test pulse signal carry out convolution algorithm are as follows:
It is limited in the time in a manner of sliding window from test and obtains audio signal fragment in the received audio signal,
Wherein, the time span of the sliding window is consistent with the test pulse signal time length, the stepping of the sliding window
Length is the integer number strong point length in window；
Convolution is done with test pulse signal to the audio signal fragment in window each in sliding process, it is involute to obtain window
The maximum value of product absolute value, and the numerical value of maximum absolute value value sign corresponding with the numerical value is saved in initially respectively
For in empty numerical value array and symbol array.
In one embodiment, the method for obtaining numerical value array and symbol array further include:
Judge whether the maximum value of convolution absolute value in the window is less than default test threshold, if being less than the default survey
Threshold value is tried, then the maximum value of the absolute value is given up, is not stored in numerical value array and symbol array.
In one embodiment, the relative time delay obtained at loudspeaker arrival reception position includes:
The greatest measure in numerical value array is obtained, institute is calculated according to position of the greatest measure in numerical value array
State the relative time delay of loudspeaker to be measured.
In one embodiment, the loudspeaker that obtains reaches the relative time delay received at position, further includes:
The corresponding sign of the greatest measure is obtained in symbol array；
The mode of connection of the loudspeaker phase to be measured is obtained according to the corresponding sign of the greatest measure.
On the other hand, the present invention provides a kind of equipment of loudspeaker delay adjust automatically, including loudspeaker to be measured and survey
Try equipment, wherein the test equipment includes:
Receiving unit, for the audio letter after the pulse signal modulation after tested of loudspeaker sending to be measured any in receiving chamber
Number；
Relative time delay computing unit, the time for issuing audio signal with the loudspeaker are the starting testing time,
It receives and convolution algorithm is carried out to the audio signal and test pulse signal that receive at position, obtain the loudspeaker and reach
Receive the relative time delay at position；
Delay parameter computing unit, it is to be measured for being calculated according to the corresponding relative time delay of all loudspeakers in interior to be measured
The corresponding delay parameter of each loudspeaker；
Output control unit, for carrying out delay tune to loudspeaker corresponding to the delay parameter according to the delay parameter
It is whole.
In one embodiment, the relative time delay computing unit includes:
Window sliding subelement, for limiting the received audio signal in the time from test in a manner of sliding window
Middle acquisition audio signal fragment, wherein the time span of the sliding window is consistent with the test pulse signal time length,
The stepping length of the sliding window is the integer number strong point length in window；
Array generates subelement, for the audio signal fragment and test pulse signal in window each in sliding process
Convolution is done, obtains the maximum value of convolution absolute value in window, and the numerical value of the maximum absolute value value is corresponding with the numerical value
Sign is saved in respectively in the numerical value array and symbol array being initially empty.
In one embodiment, the array generates subelement further include:
Threshold decision subelement, for judging whether the maximum value of convolution absolute value in the window is less than default test threshold
The maximum value of the absolute value is given up if being less than the default test threshold, is not stored in numerical value array and symbol array by value
In.
In one embodiment, the relative time delay computing unit further include:
Relative time delay obtain subelement, for according to the greatest measure in the numerical value array in the numerical value array
The relative time delay of the loudspeaker to be measured is calculated in position.
In one embodiment, the relative time delay computing unit further include:
Phase judgment sub-unit, for obtaining the loudspeaker phase to be measured according to the corresponding sign of the greatest measure
Wiring direction.
The present invention is rolled up by using the modulated audio signal issued to loudspeaker to be measured with test pulse signal
The method that product calculates realizes the purpose of large-scale indoor multiple loudspeaker delay adjust automaticallies, adjusts to solve artificial delay
Accuracy of measurement processed is not high, and the degree of automation is low, effect measures low problem, has reached automation and has realized large-scale indoor speaker delay
Adjustment, improve delay adjustment accuracy, save adjustment time cost and human cost technical effect.
Detailed description of the invention
In order to more clearly explain the embodiment of the invention or the technical proposal in the existing technology, to embodiment or will show below
There is attached drawing needed in technical description to be briefly described, it should be apparent that, the accompanying drawings in the following description is only this
Some embodiments of invention for those of ordinary skill in the art without creative efforts, can be with
It obtains other drawings based on these drawings.
Fig. 1 is a kind of method flow diagram of loudspeaker delay adjust automatically；
Fig. 2 is a kind of relative time delay acquisition process flow chart for taking loudspeaker arrival to receive at position；
Fig. 3 is a kind of equipment structure chart of loudspeaker delay adjust automatically；
Fig. 4 is a kind of structure chart of relative time delay computing unit；
Fig. 5 is the structure chart of another relative time delay computing unit；
Fig. 6 is a kind of application environment schematic diagram of specific delay adjust automatically；
Fig. 7 is a kind of specific delay adjust automatically embodiment flow chart.
Specific embodiment
Following will be combined with the drawings in the embodiments of the present invention, and technical solution in the embodiment of the present invention carries out clear, complete
Site preparation description, it is clear that described embodiments are only a part of the embodiments of the present invention, instead of all the embodiments.It is based on
Embodiment in the present invention, it is obtained by those of ordinary skill in the art without making creative efforts every other
Embodiment shall fall within the protection scope of the present invention.
In the present specification, such as adjective as first and second can be only used for by an element or movement with it is another
One element or movement distinguish, without requiring or implying any actual this relationship or sequence.In the feelings that environment allows
Under condition, one in only element, component or step should not be interpreted as limited to referring to element or component or step (s), and can
To be the one or more etc. in element, component or step.
In the present specification, for ease of description, the size of various pieces shown in the drawings is not according to actual
What proportionate relationship was drawn.
Fig. 1 is a kind of method flow schematic diagram of loudspeaker delay adjust automatically.
S11: the audio signal after the impulse modulation after tested that any loudspeaker to be measured issues in receiving chamber；
S12: being the starting testing time with the time that the loudspeaker issues audio signal, is receiving at position to receiving
The audio signal and test pulse signal carry out convolution algorithm, obtain the loudspeaker and reach receive at position opposite and prolonging
When；
S13: it is corresponding that each loudspeaker to be measured is calculated according to the corresponding relative time delay of all loudspeakers in interior to be measured
Delay parameter；
S14: delay adjustment is carried out to loudspeaker corresponding to the delay parameter according to the delay parameter.
In step s 11, before test, it is necessary first to which speaker volume is adjusted to just by the volume for adjusting loudspeaker
Required volume when often playing, here, need to illustrate when, in subsequent test process, default the volume in loudspeaker
It is not enough to cause the problem of test point is detected less than audio-frequency information.When test pulse is chosen, need to fully take into account loudspeaking
The pronunciation characteristics of device, for example, test pulse signal just needs for the loudspeaker that high and low frequency audio signal can play
Comprehensively consider this characteristic, as far as possible arrives the test all as much as possible of high and low frequency signal；Simultaneously, it is considered however that subsequent
Processing capacity, by the control of test signal pulses length in executable range.It should be noted that since general sound is believed
Number between 10ms~30ms, we can be seen to be it smoothly, for the ease of handling voice signal, be needed to receiving
Voice signal carry out adding window, once only handle the data in window, and analyzed, then remove one section of voice data again, then
It is analyzed.
A kind of common method of voice data segmentation is the method by constructing window function, that is, this window function is only at certain
There is nonzero value in one section, and the value in other sections is zero.
As one embodiment, for it can play the loudspeaker of high and low frequency audio signal, here, we select
Test pulse signal is the sinusoidal Hamming window function signal of high frequency sinusoidal signal and low frequency Hamming window product, then can test this comprehensively
Working condition of the loudspeaker in each frequency band.
Under normal circumstances, data of the high frequency sinusoidal frequency of sinusoidal Hamming window function between 6K~10KHz, in Hamming window
Point control is advisable 16~128.Here, our preferred high frequency sinusoidal signal frequencies are 8KHz, the length L of Hamming window is 32, i.e.,
Data point is 32 in Hamming window.
The data of Hamming window can be obtained by the following formula:
High frequency sinusoidal signal data can be obtained by the following formula:
Wherein, L=32 is the length of Hamming window, and n is integer.f0For the frequency of high frequency sinusoidal signal: f0=8KHz, Fs are
The sample rate of audio signal is usually arranged as 44.1KHz or 48KHz.
Furthermore, it is contemplated that may there is a situation where that phase is reversed when loudspeaker wiring, it will just for the ease of testing us
Part of the string signal data less than 0 is multiplied by coefficient 0.2, high frequency sinusoidal signal expression formula finally obtained in this way are as follows:
In conjunction with above three formula, available sine Hamming window function SINPulse expression formula are as follows:
SINPulse (n)=yysin (n) × HammData (n), 0≤n≤L-1
In addition, for the woofer that only can play low frequency audio information, in order to save the time cost of operation,
Test pulse signal only needs to consider by low frequency signal test as much as possible at this point, test pulse signal can be with
Hamming window function signal is directlyed adopt, test pulse signal can be selected as to Hamming window function signal, it may be assumed that HammData (n), together
When, the mode that selection increases Hamming window length can enable the audio signal of pulse signal modulation after tested pass through low frequency loudspeaker
Device exports out.I.e., it is possible to which the test pulse signal length of low frequency audio information, which is chosen for high and low frequency, can play test arteries and veins
2~4 times for rushing signal length.Preferably, we are by low frequency audio information Hamming window length positioning 64.
After step S11 completion, step S12 is needed to be implemented, that is, be with the time that the loudspeaker issues audio signal
The testing time is originated, convolution algorithm is carried out to the audio signal and test pulse signal that receive at position receiving, is obtained
The loudspeaker is taken to reach the relative time delay received at position.
Wherein, here specific restriction is not done to above-mentioned reception position.Reception position referred herein can be greatly
The indoor sweet spot of type is also possible to the indoor any position of tester's setting.
Specifically, it is as shown in Figure 2 to obtain the relative time delay acquisition process that the loudspeaker reaches at reception position:
S21: obtaining audio signal fragment from the audio signal in the test limitation time in a manner of sliding window,
Wherein, the time span of the sliding window is consistent with the test pulse signal time length；
S22: convolution is done with test pulse signal to the audio signal fragment in window each in sliding process, is obtained each
The maximum value of window convolution absolute value；
S23: judging whether the maximum value of convolution absolute value in the window is less than default test threshold, will be less than default survey
The maximum value for trying the absolute value of threshold value is given up, and the maximum absolute value value not less than default test threshold is corresponding with the maximum value
Sign is stored in respectively in the numerical value array and symbol array being initially empty；
S24: judge whether numerical value array is sky array；
S25: if numerical value array is empty array, it is considered as test crash, determines that the loudspeaker to be measured is faulty equipment；
S26: if numerical value array is non-empty array, the greatest measure in numerical value array is obtained, according to the greatest measure
The relative time delay of the loudspeaker to be measured is calculated in position in above-mentioned numerical value array；
S27: the corresponding sign of the greatest measure is obtained in symbol array；
S28: it if the corresponding symbol of the greatest measure is positive, establishes and saves the relative time delay and loudspeaker to be measured
One-to-one relationship；
S29: if the corresponding symbol of the greatest measure is negative, prompting phase reversed, establishes simultaneously after connecing just to phase
Save the one-to-one relationship of the relative time delay Yu loudspeaker to be measured.
Specifically, in the step s 21, being obtained from the audio signal in the test limitation time in a manner of sliding window
Take audio signal fragment.Wherein, the above-mentioned test limitation time is freely to be set by tester according to large-scale indoor concrete condition
Fixed.Preferably, a kind of test of reference limits time LTIME setting means are as follows:
In formula, vsIndicate the aerial spread speed of sound, generally 340m/s, Fs are sample rate, RoomLength
Indicate the length of longest edge in test cabinet, unit m.
The audio signal in the test limitation time is obtained in a manner of sliding window, sliding window size is test pulse letter
Number length, the stepping of sliding window are integer number strong point, it is preferred that for can play the loudspeaker of high and low frequency audio sound,
Above-mentioned window stepping is a data point, and for only can play the loudspeaker of low frequency audio sound, the stepping of above-mentioned window can be with
Extension appropriate, for example it is increased to 2 or 4 data points.
S22 calculates the convolution of audio signal fragment and test pulse signal in each sliding window.Here, for that can broadcast
The loudspeaker of high and low frequency audio-frequency information is put, test pulse signal is sinusoidal Hamming window function data-signal, low for only can play
The loudspeaker of frequency domain audio information, test pulse signal are Hamming window function data-signal.Meanwhile it is exhausted to obtain convolution in each window
To the maximum value of value.
Then, S23 is executed, judges whether the maximum value of convolution absolute value in the window is less than default test threshold；
If being less than default test threshold, the maximum value of the convolution absolute value is given up；It can determine that and do not test modulation
Audio signal afterwards, the audio-frequency information tested may be noise；
If not less than default test threshold, by the numerical value of convolution maximum absolute value value sign corresponding with the numerical value
It is stored in numerical value array and symbol array respectively.
By above-mentioned judgement, by the sign of the maximum absolute value value MAXVAL and the maximum value after judgement
MAXSIGN is stored in respectively in numerical value array MAX and symbol array SIGN, it may be assumed that
MAX=[MAXV AL1,MAXV AL2,…MAXV ALm]
SIGN=[MAXSIGN1,MAXSIGN2,…MAXSIGNm]
Then further judge whether numerical value array MAX is sky array；
If numerical value array MAX is empty array, it is considered as and does not test the audio signal that loudspeaker to be measured issues, label should
Loudspeaker is faulty equipment；
If numerical value array MAX is non-empty array, it is considered as test and has arrived the audio signal that loudspeaker to be measured issues, then,
Search for the greatest measure MAXV AL in the numerical value arrayi, and according to MAXV ALiThis is calculated in position i in array MAX
The relative time delay of loudspeaker audio signal to be measured.
Wherein, the setting for presetting test threshold can be set according to the actual situation.Specifically, for can play it is high,
For the loudspeaker of low-frequency audio signal, the setting of default test threshold (THRE) can refer to following two formula to obtain:
THRE=0.5 × 10 × MSR
In formula, when MeanSPL is loudspeaker normal play to be measured, the average electricity that data can be collected at position is being received
It is flat；MSR be loudspeaker normal play to be measured when, receive position at can collected audio data level size root mean square.
However, for be only capable of play low-frequency audio signal woofer for, can raising test threshold appropriate,
THRE × 4 × A such as is set by test threshold, wherein A is woofer test pulse length and can play high and low sound audio
The ratio of the ventional loudspeakers test pulse length of signal.
It is preferably a kind of that institute is calculated according to position of the greatest measure in above-mentioned numerical value array in step S28
State the calculation method of the relative time delay of loudspeaker to be measured are as follows:
Wherein, since TIMECOUNT represent originating the testing time, to test limitation by the convolutional calculation time in the time
Number, the i.e. number of the data point of convolutional calculation；M is total element number in numerical value array, and i represents maximum value in numerical value array
MAXV ALiThe position at place.
Further, the corresponding sign of above-mentioned greatest measure is obtained in symbol array, it is corresponding according to the greatest measure
Sign obtain the wiring direction of the loudspeaker phase to be measured.If symbol corresponding to greatest measure is negative, prompting should
The phase of loudspeaker to be measured is reversed, after needing be adjusted reversed to the phase of the loudspeaker, and saves the loudspeaker to be measured
Relative time delay value；If symbol corresponding to greatest measure is positive, the relative time delay value of the loudspeaker to be measured is saved.
Indoor all loudspeakers to be measured are successively carried out with the acquisition of above-mentioned relative time delay value, then, to be measured is raised according to all
The relative time delay of sound device obtains the delay parameter of each loudspeaker to be measured, and specific calculation method is as follows:
All loudspeakers to be measured are tested first, and obtain the corresponding relative time delay of each loudspeaker；
Search the delay maximum value MAXDELAY in all relative time delays；
Then, the corresponding delay parameter of loudspeaker of serial number k to be measured are as follows:
DELAY (k)=MAXDELAY-RDELAY (k)
In formula, RDELAY (k) represents the relative time delay of the loudspeaker of serial number k to be measured.
After the completion of delay parameter calculates, the test result of all loudspeakers to be measured is exported, and automatically according to delay parameter meter
Calculate the delay parameter that result adjusts each loudspeaker to be measured.
In the specific implementation process, the quick adjustment of speaker delay in room can be realized using the program, it is average each
The testing time of speaker can be controlled within 2 seconds, and measurement accuracy can reach within 0.5ms, was greatly saved human cost and time.
On the other hand, the present invention also provides a kind of equipment of loudspeaker delay adjust automatically, as shown in Figure 3, comprising:
Receiving unit 31, for the audio after the pulse signal modulation after tested of loudspeaker sending to be measured any in receiving chamber
Signal；
Relative time delay computing unit 32, the time for issuing audio signal with the loudspeaker are the starting testing time,
Convolution algorithm is carried out to the audio signal and test pulse signal that receive at position receiving, the loudspeaker is obtained and arrives
Up to the relative time delay received at position；
Delay parameter computing unit 33, for according to the calculating of all loudspeakers in interior to be measured corresponding relative time delay to
Survey the corresponding delay parameter of each loudspeaker；
Output control unit 34, for being delayed according to the delay parameter to loudspeaker corresponding to the delay parameter
Adjustment.
During loudspeaker is delayed adjust automatically, first before the adjustment of loudspeaker automatic time delay starts, by loudspeaker
Volume adjustment is to normal play volume, while then the default test threshold values of artificial setting is believed according to the channel configuration of loudspeaker
The playable audio frequency information of breath, i.e. loudspeaker chooses suitable test pulse signal, then loudspeaker to be measured is allowed successively to issue
Audio signal after the test pulse signal modulation, while the sending time of the audio signal is recorded, when by above-mentioned sending
Between be set as initial test period.
Specifically, the equipment of loudspeaker delay adjust automatically is placed on the position of audio signal reception.Preferably, the reception
Position be it is fixed, i.e., after being modulated by test equipment automatic time delay, finally may make indoor all loudspeakers to be measured while sending out
Sound wave after sound reaches simultaneously receives position, i.e. position where the equipment of loudspeaker delay adjust automatically.
The receiving unit of the equipment of loudspeaker delay adjust automatically is to be used to receive that above-mentioned loudspeaker to be measured to issue through surveying
Audio-frequency information after trying pulse signal modulation.
The time that relative time delay computing unit is used to issue audio signal with the loudspeaker is to originate the testing time, is being connect
It receives and convolution algorithm is carried out to the audio signal and test pulse signal that receive at position, obtain the loudspeaker arrival and connect
Receive the relative time delay at position.Specifically, relative time delay computing unit 32 further includes subelement as shown in Figure 4:
Window sliding subelement 41, for the received audio to be believed out of test the limitation time in a manner of sliding window
Audio signal fragment is obtained in number, wherein the time span of the sliding window and the test pulse signal time length one
It causes；
Array generates subelement 42, for doing convolution with test pulse signal to the audio signal fragment in window, obtains
The maximum value of convolution absolute value in window, and the numerical value of maximum absolute value value sign corresponding with the numerical value is protected respectively
It is stored in numerical value array and symbol array；
Relative time delay obtain subelement 43, for according to the greatest measure in the numerical value array in the numerical value array
Position, the relative time delay of the loudspeaker to be measured is calculated；
Phase judgment sub-unit 44, for obtaining the loudspeaker phase to be measured according to the corresponding sign of the greatest measure
The wiring direction of position.
Wherein, after the audio signal that receiving unit receives that loudspeaker to be measured is sent, window sliding subelement passes through cunning
Audio signal is segmented by the mode of dynamic window, wherein the time span of the sliding window and the test pulse signal
Time span is consistent, in addition, for the audio frequency play property of loudspeaker, the stepping of sliding window in window sliding subelement
It is inconsistent, it needs to be adjusted correspondingly according to specific audio frequency feature.
Then, array generates subelement and rolls up first to the audio signal fragment in each window with test pulse signal
Product, search convolution absolutely maximum value it is resulting to save convolutional calculation in each window further according to the sliding number of window
The maximum value of absolute value and the corresponding sign of the maximum value, and it is saved in respectively in numerical value array and symbol array.
Optimization, since the audio data that receiving unit receives may be other noise signals, in order to exclude this feelings
The interference of condition, a kind of array generation subelement 42 of optimization are as shown in Figure 5, further includes:
Threshold decision subelement 51, for judging whether the maximum value of convolution absolute value in the window is less than default test
The maximum value of the absolute value is given up if being less than the default test threshold, is not stored in numerical value array and symbol array by threshold value
In.
If the maximum value for calculating resulting convolution results absolute value in a certain window is greater than or equal to default test threshold,
Think that the audio signal that the loudspeaker to be measured issues has been arrived in test, as the maximum value of convolution results absolute value is less than default test threshold
Value, then it is assumed that do not test the audio signal that the loudspeaker to be measured issues.
Then, the maximum of saved absolute value is obtained by range in test initial time to test limitation time
Maximum value in value and the sign corresponding to it, and it is saved in respectively in the numerical value array and symbol array of update.
Relative time delay obtain subelement, for according to the greatest measure in the numerical value array in the numerical value array
The relative time delay of the loudspeaker to be measured is calculated in position.
Phase judgment sub-unit, for obtaining the loudspeaker phase to be measured according to the corresponding sign of the greatest measure
Wiring direction prompt the phase of the loudspeaker to be measured reversed if symbol corresponding to greatest measure is negative, need to raise this
The phase of sound device is reversed be adjusted after, and save the relative time delay value of the loudspeaker to be measured；If symbol corresponding to greatest measure
It number is positive, then saves the relative time delay value of the loudspeaker to be measured.
Then, indoor all loudspeakers to be measured are successively carried out with the acquisition of above-mentioned relative time delay value.
Delay parameter computing unit, it is to be measured for being calculated according to the corresponding relative time delay of all loudspeakers in interior to be measured
The corresponding delay parameter of each loudspeaker；
Output control unit, for carrying out delay tune to loudspeaker corresponding to the delay parameter according to the delay parameter
It is whole.That is, the delay test result and fault condition of all loudspeakers to be measured of output, and automatically according to delay parameter calculated result tune
The delay parameter of whole each loudspeaker to be measured.
Fig. 6 show a kind of application environment of specific loudspeaker delay adjust automatically, wherein in figure, test equipment 61
It is placed on sweet spot, that is, receives the reception position of audio-frequency information, multiple sound-box devices (i.e. loudspeaker) 62 are located at test
Indoor different location.
Fig. 7 show a kind of embodiment schematic diagram of specific loudspeaker delay adjust automatically, the following institute of testing process
Show:
S71: the volume of one speaker of adjusting to normal play volume, while the manually default test threshold of setting in test equipment
Value；
S72: judging the type of the sound-box device, is woofer or common speaker；
S73: if judging the sound-box device for woofer, woofer test is executed；
S74: if judging the sound-box device for common speaker, common speaker test is executed；
S75: judge whether to test indoor all speakers, if not surveyed, return step S71 detects next speaker
Equipment；
S76: the corresponding delay parameter of indoor each speaker is calculated according to speaker testing process result；
S77: delay adjustment is carried out to the corresponding speaker of the delay parameter according to delay parameter.
Wherein, above-mentioned common speaker is the audio signal that can play high and low frequency audio-frequency information, i.e. test pulse signal needs
This feature for comprehensively considering common speaker arrives the test more as far as possible of height frequency domain.The calculating energy of processor is also considered simultaneously
Power, by the control of test signal length in executable range.Woofer is the audio signal that can not play high-frequency information, can only
The speaker for playing the audio signal of low-frequency information, in test, pulse signal directlys adopt Hamming window function, while increasing window
Length is so that pulse signal can export out by woofer.
Specifically, test pulse signal behavior is high frequency sinusoidal signal and low frequency Hamming in common speaker testing process
The sinusoidal Hamming window signal of window product, in this way can each frequency band of complete detection speaker working condition.Then, common speaker will be through
Audio-frequency information after test pulse signal modulation is sent, and the time for starting to send is recorded as the starting testing time.
Test equipment is after receiving the audio signal after the impulse modulation after tested that common speaker is sent, with sliding window
Mode obtain test equipment acquisition audio data, sliding window size be test pulse signal length, sliding window stepping
For 1 data point.
Convolution is done with test pulse signal to the audio signal fragment in window each in sliding process, it is involute to obtain window
The maximum value of product absolute value, and the maximum value of the absolute value is compared with default test threshold, if being less than default test threshold
Value, then it is assumed that do not detect the audio-frequency information that the speaker issues, give up the maximum value.
Judged according to maximum value of the above process to convolution absolute value in all windows, finally by maximum absolute value value
Numerical value sign corresponding with the numerical value be saved in the numerical value array and symbol array being initially empty respectively.
When the numerical value array finally obtained is still empty array, it is considered as test crash, the sound-box device is marked to set for failure
It is standby；
When the numerical value array finally obtained is non-empty array, the maximum value in the numerical value array is searched for, and according to signal
Relative time delay formula calculates the relative time delay of the sound-box device；
Meanwhile symbol SIGN corresponding with the maximum value is searched in symbol array and indicates the sound when SIGN is negative
Case phase is reversed, after manually by the sound-box device phase adjustment, saves the relative time delay value of the sound-box device.When SIGN is
When, instruction is successfully tested RES=1, while saving the relative time delay value of the sound-box device.
The testing process of woofer is similar with the testing process of above-mentioned common speaker, wherein has the following to need to adjust
It is whole:
1, the test pulse signal of woofer is selected as Hamming window function signal, which generally sets
2~4 times of common speaker test pulse signal length are set to, is preferably set to 2 times of common speaker test pulse length here；
2, when test equipment detects audio signal, in order to identify low frequency signal well, by bass sound
The default test threshold of case is set as 4~16 times of common speaker, since pulse length is longer, when processor computing capability is limited
When, window can be slided stepping and be increased to 2 or 4 by 1 data point.
Other testing process of woofer are consistent with common speaker, are not unfolded to repeat herein.
It can be seen from the above description that the present invention is by using the modulated audio issued to loudspeaker to be measured
The method that signal and test pulse signal carry out convolutional calculation realizes the mesh of large-scale indoor multiple loudspeaker delay adjust automaticallies
, to solve, artificial delayed modulation accuracy of measurement is not high, and the degree of automation is low, effect measures low problem, has reached certainly
Dynamicization realizes the adjustment of large-scale indoor speaker delay, improves the accuracy of delay adjustment, saves the time cost and manpower of adjustment
The technical effect of cost.
So far, the present invention has already been described in detail.In order to avoid blinding design of the invention, not to known in the field
Some details are described.Those skilled in the art as described above, disclose it can be appreciated how implementing the present invention completely
Technical solution.
Above-described specific embodiment has carried out further the purpose of the present invention, technical scheme and beneficial effects
It is described in detail, it should be understood that being not intended to limit the present invention the foregoing is merely a specific embodiment of the invention
Protection scope, all within the spirits and principles of the present invention, any modification, equivalent substitution, improvement and etc. done should all include
Within protection scope of the present invention.
Claims (10)
1. a kind of method of loudspeaker delay adjust automatically characterized by comprising
Audio signal after the impulse modulation after tested that any loudspeaker to be measured issues in receiving chamber；
It is the starting testing time with the time that the loudspeaker issues audio signal, is receiving at position to the sound received
Frequency signal and test pulse signal carry out convolution algorithm, obtain the loudspeaker and reach the relative time delay received at position；
The corresponding delay parameter of each loudspeaker to be measured is calculated according to the corresponding relative time delay of all loudspeakers in interior to be measured；
Delay adjustment is carried out to loudspeaker corresponding to the delay parameter according to the delay parameter.
2. the method according to claim 1, wherein the audio signal and test pulse signal carry out convolution fortune
The method of calculation are as follows:
It is limited in the time in a manner of sliding window from test and obtains audio signal fragment in the received audio signal,
In, the time span of the sliding window is consistent with the test pulse signal time length, and the stepping of the sliding window is long
Degree is the integer number strong point length in window；
Convolution is done with test pulse signal to the audio signal fragment in window each in sliding process, it is exhausted to obtain convolution in window
To the maximum value of value, and the numerical value of maximum absolute value value sign corresponding with the numerical value is saved in respectively and is initially empty
Numerical value array and symbol array in.
3. according to the method described in claim 2, it is characterized in that, the method for the acquisition numerical value array and symbol array is also wrapped
It includes:
Judge whether the maximum value of convolution absolute value in the window is less than default test threshold, if being less than the default test threshold
Value, then give up the maximum value of the absolute value, be not stored in numerical value array and symbol array.
4. according to the method described in claim 3, it is characterized in that, the loudspeaker that obtains reaches prolonging at reception position relatively
When include:
Obtain the greatest measure in numerical value array, according to position of the greatest measure in numerical value array be calculated it is described to
Survey the relative time delay of loudspeaker.
5. according to the method described in claim 4, it is characterized in that, the loudspeaker that obtains reaches prolonging at reception position relatively
When, further includes:
The corresponding sign of the greatest measure is obtained in symbol array；
The mode of connection of the loudspeaker phase to be measured is obtained according to the corresponding sign of the greatest measure.
6. a kind of equipment of loudspeaker delay adjust automatically, which is characterized in that including loudspeaker to be measured and test equipment, wherein
The test equipment includes:
Receiving unit, for the audio signal after the pulse signal modulation after tested of loudspeaker sending to be measured any in receiving chamber；
Relative time delay computing unit, the time for issuing audio signal with the loudspeaker is the starting testing time, is being received
Convolution algorithm is carried out to the audio signal and test pulse signal that receive at position, the loudspeaker is obtained and reaches reception
Relative time delay at position；
Delay parameter computing unit, it is to be measured each for being calculated according to the corresponding relative time delay of all loudspeakers in interior to be measured
The corresponding delay parameter of loudspeaker；
Output control unit, for carrying out delay adjustment to loudspeaker corresponding to the delay parameter according to the delay parameter.
7. equipment according to claim 6, which is characterized in that the relative time delay computing unit includes:
Window sliding subelement obtains in the received audio signal for being limited in the time in a manner of sliding window from test
Take audio signal fragment, wherein the time span of the sliding window is consistent with the test pulse signal time length, described
The stepping length of sliding window is the integer number strong point length in window；
Array generates subelement, for rolling up to the audio signal fragment in window each in sliding process with test pulse signal
Product obtains the maximum value of convolution absolute value in window, and the numerical value of the maximum absolute value value is corresponding with the numerical value positive and negative
It number is saved in the numerical value array and symbol array being initially empty respectively.
8. equipment according to claim 7, which is characterized in that the array generates subelement further include:
Threshold decision subelement, for judging whether the maximum value of convolution absolute value in the window is less than default test threshold,
If being less than the default test threshold, the maximum value of the absolute value is given up, is not stored in numerical value array and symbol array.
9. equipment according to claim 8, which is characterized in that the relative time delay computing unit further include:
Relative time delay obtains subelement, for the position according to the greatest measure in the numerical value array in the numerical value array
It sets, the relative time delay of the loudspeaker to be measured is calculated.
10. equipment according to claim 9, which is characterized in that the relative time delay computing unit further include:
Phase judgment sub-unit, for obtaining connecing for the loudspeaker phase to be measured according to the corresponding sign of the greatest measure
Line direction.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201910089278.3A CN109862503B (en) | 2019-01-30 | 2019-01-30 | Method and equipment for automatically adjusting loudspeaker delay |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201910089278.3A CN109862503B (en) | 2019-01-30 | 2019-01-30 | Method and equipment for automatically adjusting loudspeaker delay |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN109862503A true CN109862503A (en) | 2019-06-07 |
| CN109862503B CN109862503B (en) | 2021-02-23 |
Family
ID=66896714
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201910089278.3A Active CN109862503B (en) | 2019-01-30 | 2019-01-30 | Method and equipment for automatically adjusting loudspeaker delay |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN109862503B (en) |
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN111526455A (en) * | 2020-05-21 | 2020-08-11 | 菁音电子科技（上海）有限公司 | Correction enhancement method and system for vehicle-mounted sound |
| CN113993057A (en) * | 2021-10-25 | 2022-01-28 | 浙江德清知路导航科技有限公司 | Sound field self-adaption system, method and storage medium based on audio real-time positioning technology |
| CN114071315A (en) * | 2021-11-04 | 2022-02-18 | 深圳Tcl新技术有限公司 | Audio processing method and device, electronic equipment and storage medium |
Citations (17)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2003216154A (en) * | 2002-01-18 | 2003-07-30 | Kenwood Corp | Acoustic field adjusting device |
| CN101321201A (en) * | 2007-06-06 | 2008-12-10 | 大唐移动通信设备有限公司 | Echo elimination device, communication terminal and method for confirming echo delay time |
| CN101448186A (en) * | 2007-11-26 | 2009-06-03 | 鸿富锦精密工业（深圳）有限公司 | System and method for automatic regulating sound effect of a loudspeaker |
| TW200926875A (en) * | 2007-12-07 | 2009-06-16 | Hon Hai Prec Ind Co Ltd | System and method for automatic adjusting sound of speakers |
| CN101938686A (en) * | 2010-06-24 | 2011-01-05 | 中国科学院声学研究所 | Measurement system and measurement method for head-related transfer function in common environment |
| US20120121114A1 (en) * | 2007-04-11 | 2012-05-17 | Karsten Bo Rasmussen | Hearing instrument with linearized output stage |
| CN103414980A (en) * | 2013-07-30 | 2013-11-27 | 浙江中科电声研发中心 | Method for compensating delay and frequency response characteristics of multi-output channel sound system |
| CN103718574A (en) * | 2011-07-28 | 2014-04-09 | 汤姆逊许可公司 | Audio calibration system and method |
| CN104937955A (en) * | 2013-01-24 | 2015-09-23 | 杜比实验室特许公司 | Automatic loudspeaker polarity detection |
| CN106331951A (en) * | 2016-08-18 | 2017-01-11 | 苏州上声电子有限公司 | Harmonic distortion measurement method and system based on continuous logarithmic swept response |
| CN106686520A (en) * | 2017-01-03 | 2017-05-17 | 南京地平线机器人技术有限公司 | Multi-channel audio system capable of tracking user and equipment with multi-channel audio system |
| CN107409266A (en) * | 2015-02-26 | 2017-11-28 | 安特卫普大学 | Determine the computer program and method of individuation head-related transfer function and interaural difference function |
| CN107785026A (en) * | 2017-10-18 | 2018-03-09 | 会听声学科技（北京）有限公司 | A kind of delay time estimation method eliminated for set top box indoor echo |
| EP3300389A1 (en) * | 2016-09-26 | 2018-03-28 | STMicroelectronics (Research & Development) Limited | A speaker system and method |
| CN109102821A (en) * | 2018-09-10 | 2018-12-28 | 苏州思必驰信息科技有限公司 | Delay time estimation method, system, storage medium and electronic equipment |
| CN109215667A (en) * | 2017-06-29 | 2019-01-15 | 华为技术有限公司 | Delay time estimation method and device |
| CN109246570A (en) * | 2018-08-29 | 2019-01-18 | 北京声智科技有限公司 | The device and method of microphone quality inspection |
-
2019
- 2019-01-30 CN CN201910089278.3A patent/CN109862503B/en active Active
Patent Citations (17)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2003216154A (en) * | 2002-01-18 | 2003-07-30 | Kenwood Corp | Acoustic field adjusting device |
| US20120121114A1 (en) * | 2007-04-11 | 2012-05-17 | Karsten Bo Rasmussen | Hearing instrument with linearized output stage |
| CN101321201A (en) * | 2007-06-06 | 2008-12-10 | 大唐移动通信设备有限公司 | Echo elimination device, communication terminal and method for confirming echo delay time |
| CN101448186A (en) * | 2007-11-26 | 2009-06-03 | 鸿富锦精密工业（深圳）有限公司 | System and method for automatic regulating sound effect of a loudspeaker |
| TW200926875A (en) * | 2007-12-07 | 2009-06-16 | Hon Hai Prec Ind Co Ltd | System and method for automatic adjusting sound of speakers |
| CN101938686A (en) * | 2010-06-24 | 2011-01-05 | 中国科学院声学研究所 | Measurement system and measurement method for head-related transfer function in common environment |
| CN103718574A (en) * | 2011-07-28 | 2014-04-09 | 汤姆逊许可公司 | Audio calibration system and method |
| CN104937955A (en) * | 2013-01-24 | 2015-09-23 | 杜比实验室特许公司 | Automatic loudspeaker polarity detection |
| CN103414980A (en) * | 2013-07-30 | 2013-11-27 | 浙江中科电声研发中心 | Method for compensating delay and frequency response characteristics of multi-output channel sound system |
| CN107409266A (en) * | 2015-02-26 | 2017-11-28 | 安特卫普大学 | Determine the computer program and method of individuation head-related transfer function and interaural difference function |
| CN106331951A (en) * | 2016-08-18 | 2017-01-11 | 苏州上声电子有限公司 | Harmonic distortion measurement method and system based on continuous logarithmic swept response |
| EP3300389A1 (en) * | 2016-09-26 | 2018-03-28 | STMicroelectronics (Research & Development) Limited | A speaker system and method |
| CN106686520A (en) * | 2017-01-03 | 2017-05-17 | 南京地平线机器人技术有限公司 | Multi-channel audio system capable of tracking user and equipment with multi-channel audio system |
| CN109215667A (en) * | 2017-06-29 | 2019-01-15 | 华为技术有限公司 | Delay time estimation method and device |
| CN107785026A (en) * | 2017-10-18 | 2018-03-09 | 会听声学科技（北京）有限公司 | A kind of delay time estimation method eliminated for set top box indoor echo |
| CN109246570A (en) * | 2018-08-29 | 2019-01-18 | 北京声智科技有限公司 | The device and method of microphone quality inspection |
| CN109102821A (en) * | 2018-09-10 | 2018-12-28 | 苏州思必驰信息科技有限公司 | Delay time estimation method, system, storage medium and electronic equipment |
Non-Patent Citations (1)
| Title |
|---|
| 段云炀: "《基于谱矩阵分解重构的波束形成声源识别法》", 《中国优秀硕士学位论文全文数据库工程科技Ⅱ辑》 * |
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN111526455A (en) * | 2020-05-21 | 2020-08-11 | 菁音电子科技（上海）有限公司 | Correction enhancement method and system for vehicle-mounted sound |
| CN113993057A (en) * | 2021-10-25 | 2022-01-28 | 浙江德清知路导航科技有限公司 | Sound field self-adaption system, method and storage medium based on audio real-time positioning technology |
| CN114071315A (en) * | 2021-11-04 | 2022-02-18 | 深圳Tcl新技术有限公司 | Audio processing method and device, electronic equipment and storage medium |
Also Published As
| Publication number | Publication date |
|---|---|
| CN109862503B (en) | 2021-02-23 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| US8798280B2 (en) | Calibration method and device in an audio system | |
| CN109862503A (en) | A kind of method and apparatus of loudspeaker delay adjust automatically | |
| US8175284B2 (en) | Method and apparatus for calibrating sound-reproducing equipment | |
| CN109275084A (en) | Test method, device, system, equipment and the storage medium of microphone array | |
| CN108235181B (en) | Method for noise reduction in an audio processing apparatus | |
| CN103915103A (en) | Voice quality enhancement system | |
| CN105277921B (en) | A kind of passive acoustic localization method based on smart mobile phone | |
| WO2004075167A2 (en) | Log-likelihood ratio method for detecting voice activity and apparatus | |
| CN106612482A (en) | Method for adjusting audio parameter and mobile terminal | |
| CN109658935B (en) | Method and system for generating multi-channel noisy speech | |
| CN107708048A (en) | Detection method of uttering long and high-pitched sounds and device, storage medium and electronic installation | |
| CN109040911B (en) | Intelligent sound box and determination method for target placement position thereof | |
| CN109672966A (en) | A kind of voice pick-up method, device and system | |
| CN112037816B (en) | Correction, howling detection and suppression method and device for frequency domain frequency of voice signal | |
| CN110139204A (en) | Intelligent sound equipment acoustical behavior test method and system | |
| WO2017201868A1 (en) | Method, apparatus and terminal for adjusting speech gain | |
| CN109195090A (en) | Test method and system for microphone electro mechanical parameter in product | |
| WO2005104610A1 (en) | Resonance frequency determining method, resonance frequency selecting method, and resonance frequency determining apparatus | |
| CN108174337A (en) | A kind of adaptive method of room sound field and combination loudspeaker system | |
| CN110267155A (en) | A kind of control method and speaker of speaker | |
| CN105791530B (en) | Output volume adjusting method and apparatus | |
| CN108605191B (en) | Abnormal sound detection method and device | |
| CN109195089A (en) | Test method and system for loudspeaker electro mechanical parameter in product | |
| WO2023051622A1 (en) | Method for improving far-field speech interaction performance, and far-field speech interaction system | |
| CN113424558A (en) | Intelligent personal assistant |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| PB01 | Publication | ||
| PB01 | Publication | ||
| SE01 | Entry into force of request for substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| GR01 | Patent grant | ||
| GR01 | Patent grant |