CN109862503B - Method and equipment for automatically adjusting loudspeaker delay - Google Patents
Method and equipment for automatically adjusting loudspeaker delay Download PDFInfo
- Publication number
- CN109862503B CN109862503B CN201910089278.3A CN201910089278A CN109862503B CN 109862503 B CN109862503 B CN 109862503B CN 201910089278 A CN201910089278 A CN 201910089278A CN 109862503 B CN109862503 B CN 109862503B
- Authority
- CN
- China
- Prior art keywords
- loudspeaker
- delay
- value
- array
- test
- 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.)
- Active
Links
Images
Landscapes
- Stereophonic System (AREA)
Abstract
The invention provides a method and equipment for automatically adjusting a loudspeaker delay parameter, wherein the method comprises the following steps: receiving an audio signal which is sent by any loudspeaker to be tested and modulated by a test pulse; taking the time of sending the audio signal by the loudspeaker as the initial test time, carrying out convolution operation on the received audio signal and the test pulse signal at the receiving position, and obtaining the relative delay of the loudspeaker reaching the receiving position; calculating a corresponding delay parameter of each loudspeaker to be tested according to the corresponding relative delay of all the loudspeakers in the room to be tested; and carrying out delay adjustment on the loudspeaker corresponding to the delay parameter according to the delay parameter. By the scheme, the problem of delay adjustment of the large indoor loudspeaker is solved, and the technical effects of saving the labor cost and time cost of delay adjustment and improving the accuracy of delay adjustment can be achieved.
Description
Technical Field
The invention relates to the field of voice signal processing, in particular to a method and equipment for automatically adjusting loudspeaker time delay.
Background
At the cinema, the conference room, large-scale indoor occasion such as stage and the large-scale booth of KTV, after new project construction is accomplished, because indoor occasion is great, often there are a plurality of speakers to place far away from, and differ from listening position, the sound information that leads to a plurality of speakers to arrive listening position after the simultaneous sound production has different delays, this not only can produce phase deviation, the intensity that leads to sound information reduces, and because the stack of different delays, can lead to sound information's definition to descend, very big reduction listening effect and user experience.
In general, engineers may set different delay parameters for each speaker in a large room by experience using methods such as visual estimation and distance measurement, so as to make the delay of sound information emitted by each speaker in the room to reach the listening position correspond. For example, some engineers may use an infrared range finder to measure the distance between each speaker and the listening location, and then calculate the delay from the sound information emitted from each speaker to the listening location through a sound propagation velocity formula, and further calculate the delay that each speaker needs to arrive at the listening location at the same time. The manual measurement method has the problems of low measurement accuracy, low automation degree, low efficiency and low delay accuracy, and engineering personnel are required to calculate delay by themselves.
At present, no technical scheme capable of effectively solving the problems is provided for solving the problems.
Disclosure of Invention
The invention provides a method and equipment for automatically adjusting loudspeaker time delay, which realize the aim of automatically adjusting the time delay of a plurality of loudspeakers in a large-scale room, achieve the technical effects of automatically adjusting the time delay of a sound box in the large-scale room, improving the accuracy of time delay adjustment and saving the time cost and the labor cost of adjustment.
In one aspect, the present invention provides a method for automatically adjusting speaker delay, including:
receiving an audio signal which is sent by any loudspeaker to be tested and modulated by a test pulse;
taking the time of sending the audio signal by the loudspeaker as the initial test time, carrying out convolution operation on the received audio signal and the test pulse signal at the receiving position, and obtaining the relative delay of the loudspeaker reaching the receiving position;
calculating a corresponding delay parameter of each loudspeaker to be tested according to the corresponding relative delay of all the loudspeakers in the room to be tested;
and carrying out delay adjustment on the loudspeaker corresponding to the delay parameter according to the delay parameter.
In one embodiment, the convolution operation of the audio signal and the test pulse signal is performed by:
acquiring an audio signal segment from the audio signal received within the test limit time in a sliding window mode, wherein the time length of the sliding window is consistent with the time length of the test pulse signal, and the step length of the sliding window is an integer number of data point lengths within the window;
and convolving the audio signal segment in each window in the sliding process with the test pulse signal to obtain the maximum value of the convolution absolute value in the window, and respectively storing the value of the maximum value of the absolute value and the sign corresponding to the value into an initially empty value array and a symbol array.
In one embodiment, the method of obtaining the value array and the symbol array further includes:
and judging whether the maximum value of the convolution absolute value in the window is smaller than a preset test threshold value or not, if so, discarding the maximum value of the absolute value and not storing the maximum value into a numerical value array and a symbol array.
In one embodiment, the obtaining the relative delay time of the arrival of the speaker at the receiving position comprises:
and acquiring the maximum value in the numerical array, and calculating according to the position of the maximum value in the numerical array to obtain the relative delay of the loudspeaker to be tested.
In one embodiment, the obtaining the relative delay time of the arrival of the speaker at the receiving position further comprises:
acquiring the sign corresponding to the maximum numerical value in a symbol array;
and obtaining the wiring method of the phase of the loudspeaker to be tested according to the sign corresponding to the maximum value.
In another aspect, the present invention provides an apparatus for automatically adjusting speaker delay, including a speaker to be tested and a testing apparatus, wherein the testing apparatus includes:
the receiving unit is used for receiving the audio signal which is sent by any loudspeaker to be tested and modulated by the test pulse signal;
the relative delay calculating unit is used for carrying out convolution operation on the received audio signal and the test pulse signal at a receiving position by taking the time of sending the audio signal by the loudspeaker as the initial test time to obtain the relative delay of the loudspeaker reaching the receiving position;
the delay parameter calculating unit is used for calculating the delay parameter corresponding to each loudspeaker to be detected according to the relative delay corresponding to all the loudspeakers in the room to be detected respectively;
and the output control unit is used for carrying out delay adjustment on the loudspeaker corresponding to the delay parameter according to the delay parameter.
In one embodiment, the relative delay calculating unit includes:
the window sliding subunit is used for acquiring an audio signal segment from the audio signal received within the test limit time in a sliding window mode, wherein the time length of the sliding window is consistent with the time length of the test pulse signal, and the step length of the sliding window is an integer number of data point lengths within the window;
and the array generating subunit is used for convolving the audio signal segment in each window with the test pulse signal in the sliding process, acquiring the maximum value of the convolution absolute value in the window, and respectively storing the value of the maximum value of the absolute value and the sign corresponding to the value into an initially empty value array and a sign array.
In one embodiment, the array generation subunit further comprises:
and the threshold judgment subunit is used for judging whether the maximum value of the convolution absolute value in the window is smaller than a preset test threshold, and if the maximum value of the convolution absolute value in the window is smaller than the preset test threshold, the maximum value of the absolute value is discarded and is not stored in the numerical value array and the symbol array.
In one embodiment, the relative delay calculating unit further includes:
and the relative delay obtaining subunit is used for calculating the relative delay of the loudspeaker to be tested according to the position of the maximum numerical value in the numerical value array.
In one embodiment, the relative delay calculating unit further includes:
and the phase judgment subunit is used for acquiring the wiring direction of the phase of the loudspeaker to be tested according to the sign corresponding to the maximum value.
The invention realizes the purpose of automatically adjusting the time delay of a plurality of loudspeakers in a large room by adopting the method of carrying out convolution calculation on the modulated audio signal sent by the loudspeaker to be tested and the test pulse signal, thereby solving the problems of low measurement accuracy, low automation degree and low effective amount of manual time delay modulation, achieving the technical effects of automatically adjusting the time delay of a large-scale indoor sound box, improving the accuracy of time delay adjustment and saving the time cost and the labor cost of adjustment.
Drawings
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings used in the description of the embodiments or the prior art will be briefly described below, it is obvious that the drawings in the following description are only some embodiments of the present invention, and for those skilled in the art, other drawings can be obtained according to the drawings without creative efforts.
FIG. 1 is a flow chart of a method for automatic adjustment of speaker delay;
fig. 2 is a flow chart of a relative delay acquisition process for acquiring the arrival of a speaker at a receiving position;
FIG. 3 is a block diagram of an apparatus for automatic adjustment of speaker delay;
FIG. 4 is a block diagram of a relative delay calculation unit;
FIG. 5 is a block diagram of still another relative delay calculating unit;
FIG. 6 is a diagram of an exemplary application environment for automatic delay adjustment;
fig. 7 is a flowchart of an embodiment of automatic delay adjustment.
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.
In this specification, adjectives such as first and second may only be used to distinguish one element or action from another, without necessarily requiring or implying any actual such relationship or order. References to an element or component or step (etc.) should not be construed as limited to only one of the element, component, or step, but rather to one or more of the element, component, or step, etc., where the context permits.
In the present specification, the sizes of the respective portions shown in the drawings are not drawn in an actual proportional relationship for the convenience of description.
Fig. 1 is a schematic flow chart of a method for automatically adjusting the time delay of a speaker.
S11: receiving an audio signal which is sent by any loudspeaker to be tested and modulated by a test pulse;
s12: taking the time of sending the audio signal by the loudspeaker as the initial test time, carrying out convolution operation on the received audio signal and the test pulse signal at the receiving position, and obtaining the relative delay of the loudspeaker reaching the receiving position;
s13: calculating a corresponding delay parameter of each loudspeaker to be tested according to the corresponding relative delay of all the loudspeakers in the room to be tested;
s14: and carrying out delay adjustment on the loudspeaker corresponding to the delay parameter according to the delay parameter.
In step S11, before the test, the volume of the speaker needs to be adjusted to the volume required for normal playing, where, when it needs to be described, the volume of the speaker is not enough to cause the problem that no audio information can be detected at the test point by default in the subsequent test process. When the test pulse is selected, the pronunciation characteristics of the speaker need to be fully considered, for example, for the speaker capable of playing both high and low frequency audio signals, the test pulse signal needs to comprehensively consider the characteristics, and the high and low frequency signals are tested as much as possible; meanwhile, the pulse length of the test signal is controlled within the executable range in consideration of the subsequent processing capability. It should be noted that, because the general sound signal is between 10ms and 30ms, it can be seen as stable, in order to process the sound signal, it is necessary to window the received sound signal, process only the data in the window once, analyze the data, then take down a segment of sound data, and analyze the data.
A common method of sound data segmentation is by constructing a window function, i.e. the window function has non-zero values only in some intervals, while values in other intervals are zero.
As an embodiment, for a speaker capable of playing high and low frequency audio signals, we select the test pulse signal as a sine hamming window function signal of the product of the high frequency sine signal and the low frequency hamming window, so that the operation of the speaker in each frequency band can be fully tested.
Generally, the high-frequency sine frequency of the sine Hamming window function is between 6K and 10KHz, and the data points in the Hamming window are preferably controlled to be between 16 and 128. Here, we prefer the high frequency sinusoidal signal frequency to be 8KHz and the hamming window length L to be 32, i.e. 32 data points within the hamming window.
The data for the hamming window can be obtained from the following equation:
the high frequency sinusoidal signal data may be obtained by the following equation:
wherein, L is 32, which is the length of Hamming window, and n is integer. f. of0Frequency of the high frequency sinusoidal signal: f. of0Fs is the sampling rate of the audio signal, typically set to 44.1KHz or 48 KHz.
In addition, considering that the situation that the phase is reversed when the loudspeaker is connected, for the convenience of test, the part of the sinusoidal signal data which is less than 0 is multiplied by a coefficient of 0.2, so that the expression of the finally obtained high-frequency sinusoidal signal is as follows:
by combining the three formulas, the expression of the sine hamming window function SINPulse can be obtained as follows:
SINPulse(n)＝yysin(n)×HammData(n),0≤n≤L-1
in addition, for a low-frequency speaker capable of playing only low-frequency audio information, in order to save the time cost of operation, the test pulse signal only needs to consider to test the low-frequency signal as much as possible, at this time, the test pulse signal may directly adopt a hamming window function signal, and the test pulse signal may be selected as a hamming window function signal, that is: hammdata (n), and meanwhile, the audio signal modulated by the test pulse signal can be output through the low-frequency loudspeaker by selecting a mode of increasing the length of the Hamming window. That is, the length of the test pulse signal of the low-frequency audio information can be selected to be 2 to 4 times of the length of the test pulse signal which can be played by both high and low frequencies. Preferably, we locate 64 the hamming window length for low frequency audio information.
After step S11 is completed, step S12 is executed, that is, taking the time of the audio signal emitted by the speaker as the starting test time, the received audio signal is convolved with the test pulse signal at the receiving position, and the relative delay of the speaker reaching the receiving position is obtained.
Here, the receiving position is not specifically limited. The receiving location referred to herein may be the sweet spot in a large room or any location in the room that is set by the tester.
Specifically, the process of acquiring the relative delay from the speaker to the receiving position is shown in fig. 2:
s21: acquiring an audio signal segment from the audio signal within a test limit time in a sliding window mode, wherein the time length of the sliding window is consistent with the time length of the test pulse signal;
s22: convolving the audio signal segment in each window with the test pulse signal in the sliding process to obtain the maximum value of the convolution absolute value of each window;
s23: judging whether the maximum value of the convolution absolute value in the window is smaller than a preset test threshold value or not, discarding the maximum value of the absolute value smaller than the preset test threshold value, and respectively storing the maximum value of the absolute value not smaller than the preset test threshold value and the sign corresponding to the maximum value in an initially empty numerical value array and a sign array;
s24: judging whether the numerical array is a null array or not;
s25: if the numerical array is a null array, the test is regarded as failed, and the loudspeaker to be tested is judged to be fault equipment;
s26: if the numerical array is a non-null array, acquiring the maximum numerical value in the numerical array, and calculating according to the position of the maximum numerical value in the numerical array to obtain the relative delay of the loudspeaker to be tested;
s27: acquiring the sign corresponding to the maximum numerical value in a symbol array;
s28: if the symbol corresponding to the maximum numerical value is positive, establishing and storing a one-to-one corresponding relation between the relative delay and the loudspeaker to be tested;
s29: and if the sign corresponding to the maximum numerical value is negative, prompting that the phase is reversely connected, and establishing and storing the one-to-one corresponding relation between the relative delay and the loudspeaker to be tested after the phase is positively connected.
Specifically, in step S21, an audio signal segment is obtained from the audio signal within the test limit time in a sliding window manner. Wherein, the test limit time is freely set by a tester according to the specific situation in a large-scale room. Preferably, a reference test limit time LTIME setting manner is:
in the formula, vsWhich represents the speed of sound propagation in air and is typically 340m/s, Fs is the sampling rate, and RoomLength represents the length of the longest side in the test chamber in m.
The audio signal within the test limit time is obtained in a sliding window manner, the size of the sliding window is the length of the test pulse signal, the step of the sliding window is an integer number of data points, preferably, for a loudspeaker capable of playing high-frequency and low-frequency audio sounds, the window is stepped to one data point, and for a loudspeaker capable of playing only low-frequency audio sounds, the step of the window can be appropriately prolonged, for example, increased to 2 or 4 data points.
S22, calculating the convolution of the audio signal segment and the test pulse signal within each sliding window. Here, the test pulse signal is a sinusoidal hamming window function data signal for a speaker capable of playing high and low frequency audio information, and the test pulse signal is a hamming window function data signal for a speaker capable of playing only low frequency audio information. Meanwhile, the maximum value of the absolute value of the convolution in each window is obtained.
Then, executing S23, and judging whether the maximum value of the convolution absolute values in the window is smaller than a preset test threshold value;
if the maximum value of the convolution absolute value is smaller than a preset test threshold value, discarding the maximum value of the convolution absolute value; the modulated audio signal can be judged not to be tested, and the tested audio information can be noise;
if the convolution absolute value is not less than the preset test threshold, the numerical value of the maximum value of the convolution absolute value and the sign corresponding to the numerical value are respectively stored in a numerical value array and a sign array.
Through the above judgment, the maximum absolute value MAXVAL and the SIGN MAXSIGN of the maximum value after the judgment are respectively stored in the numerical array MAX and the SIGN array SIGN, that is:
MAX＝[MAXV AL1,MAXV AL2,…MAXV ALm]
SIGN＝[MAXSIGN1,MAXSIGN2,…MAXSIGNm]
then further judging whether the numerical array MAX is a null array;
if the numerical array MAX is a null array, the audio signal sent by the loudspeaker to be tested is not tested, and the loudspeaker is marked as a fault device;
if the numerical array MAX is a non-null array, the audio signal emitted by the loudspeaker to be tested is considered to be tested, and then the maximum numerical value MAXV AL in the numerical array is searchediAnd according to MAXV ALiAnd calculating the relative delay of the audio signal of the loudspeaker to be tested at the position i in the array MAX.
The setting of the preset test threshold value can be set according to the actual situation. Specifically, for a speaker capable of playing high-frequency and low-frequency audio signals, the setting of the preset test threshold (three) can be obtained by referring to the following two formulas:
THRE＝0.5×10×MSR
in the formula, MeanSPL is an average level of data which can be collected at a receiving position when a speaker to be tested is normally played; the MSR is the root mean square of the level of the audio data which can be collected at the receiving position when the loudspeaker to be tested is normally played.
However, for woofers that can only play low frequency audio signals, the test threshold may be increased appropriately, such as setting the test threshold to threx 4 × a, where a is the ratio of the woofer test pulse length to the normal speaker test pulse length that can play high and low frequency audio signals.
In step S28, a preferred method for calculating the relative delay of the speaker to be tested according to the position of the maximum value in the above numerical array includes:
wherein, the TIMECOUNT represents the convolution calculation times from the initial test time to the test limit ending time, namely the number of data points of the convolution calculation; m is the total number of elements in the numerical array, and i represents the maximum value MAXV AL in the numerical arrayiThe location of the location.
Further, the sign corresponding to the maximum value is obtained from the symbol array, and the wiring direction of the phase of the loudspeaker to be tested is obtained according to the sign corresponding to the maximum value. If the sign corresponding to the maximum numerical value is negative, prompting that the phase of the loudspeaker to be tested is reversely connected, adjusting the reverse connection of the phase of the loudspeaker and storing the relative delay value of the loudspeaker to be tested; if the symbol corresponding to the maximum value is positive, the relative delay value of the loudspeaker to be tested is saved.
The relative delay values of all indoor speakers to be tested are sequentially obtained, then, the delay parameter of each speaker to be tested is obtained according to the relative delay of all speakers to be tested, and the specific calculation method is as follows:
firstly, testing all speakers to be tested, and acquiring the relative delay corresponding to each speaker;
searching a maximum time delay value MAXDELAY in all relative time delays;
then, the delay parameter corresponding to the speaker with the serial number k to be measured is:
DELAY(k)＝MAXDELAY-RDELAY(k)
in the formula, rdelay (k) represents the relative delay of the speaker with the serial number k to be measured.
And after the time delay parameter calculation is finished, outputting the test results of all the speakers to be tested, and automatically adjusting the time delay parameter of each speaker to be tested according to the time delay parameter calculation result.
In the specific implementation process, the scheme can be used for realizing the rapid adjustment of the time delay of the sound boxes in the room, the average test time of each sound box can be controlled within 2 seconds, the measurement precision can be within 0.5ms, and the labor cost and time are greatly saved.
In another aspect, the present invention further provides an apparatus for automatically adjusting a speaker delay, as shown in fig. 3, including:
the receiving unit 31 is configured to receive an audio signal modulated by a test pulse signal and emitted by any indoor speaker to be tested;
a relative delay calculating unit 32, configured to perform convolution operation on the received audio signal and the test pulse signal at a receiving position by using the time when the speaker sends the audio signal as an initial test time, and obtain a relative delay when the speaker reaches the receiving position;
the delay parameter calculating unit 33 is configured to calculate a delay parameter corresponding to each speaker to be tested according to the relative delays corresponding to all speakers in the room to be tested;
and the output control unit 34 is configured to perform delay adjustment on the speaker corresponding to the delay parameter according to the delay parameter.
In the process of automatic loudspeaker delay adjustment, firstly, before the automatic loudspeaker delay adjustment is started, the loudspeaker volume is adjusted to the normal playing volume, meanwhile, a preset test threshold value is manually set, then, according to sound channel configuration information of the loudspeaker, namely, audio frequency information which can be played by the loudspeaker, a proper test pulse signal is selected, then the loudspeaker to be tested sequentially sends out the audio signal modulated by the test pulse signal, meanwhile, the sending time of the audio signal is recorded, and the sending time is set as the initial test time.
Specifically, the device for automatically adjusting the loudspeaker delay is placed at the position where the audio signal is received. Preferably, the receiving position is fixed, that is, after the automatic delay modulation is performed by the testing device, the sound waves generated by all indoor speakers to be tested can finally reach the receiving position at the same time, that is, the position of the device for automatically adjusting the delay of the speakers is located.
And the receiving unit of the equipment for automatically adjusting the loudspeaker time delay is used for receiving the audio information which is sent by the loudspeaker to be tested and modulated by the test pulse signal.
The relative delay calculating unit is used for performing convolution operation on the received audio signal and the test pulse signal at the receiving position by taking the time of the audio signal sent by the loudspeaker as the initial test time, and acquiring the relative delay of the loudspeaker reaching the receiving position. Specifically, the relative delay calculating unit 32 further includes sub-units shown in fig. 4:
a window sliding subunit 41, configured to obtain an audio signal segment from the audio signal received within a test limit time in a sliding window manner, where a time length of the sliding window is consistent with a time length of the test pulse signal;
the array generating subunit 42 is configured to convolve the audio signal segment in the window with the test pulse signal, obtain a maximum value of an absolute value of the convolution in the window, and store a value of the maximum value of the absolute value and a sign corresponding to the value in a value array and a sign array, respectively;
a relative delay obtaining subunit 43, configured to calculate, according to a position of a maximum value in the numerical array, a relative delay of the to-be-tested speaker;
and the phase judgment subunit 44 is configured to obtain the connection direction of the phase of the speaker to be tested according to the sign corresponding to the maximum value.
After the receiving unit receives the audio signal sent by the speaker to be tested, the window sliding subunit segments the audio signal in a sliding window mode, wherein the time length of the sliding window is consistent with the time length of the test pulse signal, and in addition, for the audio frequency playing characteristic of the speaker, the step of the sliding window in the window sliding subunit is inconsistent, and corresponding adjustment needs to be performed according to specific audio frequency characteristics.
Then, the array generation subunit firstly convolutes the audio signal segment in each window with the test pulse signal, finds the maximum value of the convolution, then stores the maximum value of the absolute value obtained by convolution calculation in each window and the sign corresponding to the maximum value according to the sliding times of the windows, and stores the maximum value and the sign corresponding to the maximum value in the numerical array and the sign array respectively.
Optimally, since the audio data received by the receiving unit may be other noise signals, in order to eliminate the interference of this situation, an optimized array generating subunit 42 is shown in fig. 5, and further includes:
and a threshold judgment subunit 51, configured to judge whether the maximum value of the absolute value of the convolution in the window is smaller than a preset test threshold, and if the maximum value of the absolute value is smaller than the preset test threshold, discard the maximum value of the absolute value and store the maximum value of the absolute value into the numerical array and the symbol array.
And if the maximum value of the absolute value of the convolution result calculated in a certain window is greater than or equal to a preset test threshold value, determining that the audio signal sent by the loudspeaker to be tested is tested, and if the maximum value of the absolute value of the convolution result is less than the preset test threshold value, determining that the audio signal sent by the loudspeaker to be tested is not tested.
Then, in the range from the test starting time to the test limiting time, the maximum value of the stored maximum values of the absolute values and the corresponding sign are obtained and are respectively stored in the updated numerical value array and the updated sign array.
And the relative delay obtaining subunit is used for calculating the relative delay of the loudspeaker to be tested according to the position of the maximum numerical value in the numerical value array.
The phase judgment subunit is configured to obtain the wiring direction of the phase of the speaker to be tested according to the sign corresponding to the maximum value, prompt that the phase of the speaker to be tested is connected reversely if the sign corresponding to the maximum value is negative, and save the relative delay value of the speaker to be tested after the phase of the speaker needs to be adjusted; if the symbol corresponding to the maximum value is positive, the relative delay value of the loudspeaker to be tested is saved.
And then, acquiring the relative delay values of all the indoor speakers to be tested in sequence.
The delay parameter calculating unit is used for calculating the delay parameter corresponding to each loudspeaker to be detected according to the relative delay corresponding to all the loudspeakers in the room to be detected respectively;
and the output control unit is used for carrying out delay adjustment on the loudspeaker corresponding to the delay parameter according to the delay parameter. Namely, the delay test results and the fault conditions of all the speakers to be tested are output, and the delay parameters of each speaker to be tested are automatically adjusted according to the delay parameter calculation results.
Fig. 6 shows a specific application environment for automatic adjustment of the loudspeaker delay, in which a test device 61 is placed at the sweet spot, i.e. the receiving location for receiving audio information, and a plurality of enclosure devices (i.e. loudspeakers) 62 are located at different locations in the test room.
Fig. 7 is a schematic diagram of an embodiment of a specific automatic speaker delay adjustment, and a test procedure is as follows:
s71: adjusting the volume of a sound box to the normal playing volume, and manually setting a preset test threshold value in test equipment;
s72: judging whether the type of the sound box equipment is a bass sound box or a common sound box;
s73: if the sound box equipment is judged to be a bass sound box, executing a bass sound box test;
s74: if the sound box equipment is judged to be a common sound box, executing a common sound box test;
s75: judging whether all the sound boxes in the room are tested, if not, returning to the step S71 to detect the next sound box device;
s76: calculating a delay parameter corresponding to each indoor sound box according to the sound box test process result;
s77: and carrying out delay adjustment on the sound box corresponding to the delay parameter according to the delay parameter.
The common sound box is an audio signal capable of playing high-frequency and low-frequency audio information, namely, the high-frequency and low-frequency domains are tested as much as possible by comprehensively considering the characteristic of the common sound box for testing the pulse signal. Meanwhile, the length of the test signal is controlled within an executable range by considering the computing capacity of the processor. The bass sound box is a sound box which can not play the audio signal of high-frequency information and can only play the audio signal of low-frequency information, and during testing, the pulse signal directly adopts a Hamming window function, and simultaneously the window length is increased so that the pulse signal can be output through the bass sound box.
Specifically, in the test process of the common sound box, the test pulse signal is selected to be a sine Hamming window signal of the product of a high-frequency sine signal and a low-frequency Hamming window, so that the working condition of each frequency section of the sound box can be comprehensively detected. Then, the common sound box sends out the audio information modulated by the test pulse signal, and records the time for starting sending as the initial test time.
After receiving the audio signal modulated by the test pulse and sent by the common sound box, the test equipment acquires the audio data collected by the test equipment in a sliding window mode, wherein the size of the sliding window is the length of the test pulse signal, and the step of the sliding window is 1 data point.
And convolving the audio signal segment in each window in the sliding process with the test pulse signal to obtain the maximum value of the convolution absolute value in the window, comparing the maximum value of the absolute value with a preset test threshold value, if the maximum value is smaller than the preset test threshold value, determining that the audio information sent by the loudspeaker box is not detected, and discarding the maximum value.
And finally, respectively storing the numerical value of the maximum absolute value and the sign corresponding to the numerical value into an initially empty numerical value array and a sign array.
When the finally obtained numerical array is still a null array, the test is regarded as failed, and the sound box equipment is marked as fault equipment;
when the finally obtained numerical array is a non-empty array, searching the maximum value in the numerical array, and calculating the relative delay of the sound box equipment according to a signal relative delay formula;
and meanwhile, searching a SIGN corresponding to the maximum value in the SIGN array, indicating that the phase of the sound box is reversed when the SIGN is negative, and saving the relative delay value of the sound box equipment after manually adjusting the phase of the sound box equipment. When SIGN is yes, it indicates that the test success RES is 1 while the relative delay value of the sound box device is saved.
The testing process of the bass sound box is similar to that of the common sound box, wherein the following points need to be adjusted:
1. selecting a Hamming window function signal as a test pulse signal of the bass sound box, wherein the length of the Hamming window function signal is generally set to be 2-4 times of the length of a test pulse signal of a common sound box, and preferably set to be 2 times of the length of the test pulse signal of the common sound box;
2. when the test equipment detects the audio signal, in order to well identify the low-frequency signal, the preset test threshold of the bass sound box is set to be 4-16 times of that of a common sound box, and due to the fact that the pulse length is long, when the calculation capacity of the processor is limited, the window can be increased from 1 data point to 2 or 4 data points in a sliding stepping mode.
Other testing procedures of the bass sound box are consistent with those of the common sound box, and are not described herein.
From the above description, it can be seen that the present invention achieves the purpose of automatic adjustment of time delay of a plurality of loudspeakers in a large-scale room by adopting the convolution calculation method of the modulated audio signal emitted by the loudspeaker to be tested and the test pulse signal, thereby solving the problems of low accuracy of measurement, low automation degree and low effective amount of manual time delay modulation, and achieving the technical effects of automatically adjusting time delay of a large-scale indoor sound box, improving accuracy of time delay adjustment, and saving time cost and labor cost of adjustment.
Thus far, the present invention has been described in detail. Some details well known in the art have not been described in order to avoid obscuring the concepts of the present invention. Those skilled in the art can fully appreciate how to implement the disclosed embodiments in light of the foregoing description.
The above-mentioned embodiments are intended to illustrate the objects, technical solutions and advantages of the present invention in further detail, and it should be understood that the above-mentioned embodiments are merely exemplary embodiments of the present invention, and are not intended to limit the scope of the present invention, and any modifications, equivalent substitutions, improvements and the like made within the spirit and principle of the present invention should be included in the scope of the present invention.
Claims (8)
1. A method for automatically adjusting the time delay of a loudspeaker is characterized by comprising the following steps:
receiving an audio signal which is sent by any loudspeaker to be tested and modulated by a test pulse;
taking the time of sending the audio signal by the loudspeaker as the initial test time, carrying out convolution operation on the received audio signal and the test pulse signal at the receiving position, and obtaining the relative delay of the loudspeaker reaching the receiving position; the test pulse signals are selected according to the pronunciation characteristics of the loudspeaker;
performing convolution operation on the audio signal and the test pulse signal comprises acquiring an audio signal segment from the audio signal received within test limit time in a sliding window mode, wherein the time length of the sliding window is consistent with the time length of the test pulse signal, and the step length of the sliding window is an integer number of data point lengths within the window;
convolving the audio signal segment in each window in the sliding process with the test pulse signal to obtain the maximum value of the convolution absolute value in the window, and respectively storing the value of the maximum value of the absolute value and the sign corresponding to the value into an initially empty value array and a symbol array;
calculating a corresponding delay parameter of each loudspeaker to be tested according to the corresponding relative delay of all the loudspeakers in the room to be tested;
and carrying out delay adjustment on the loudspeaker corresponding to the delay parameter according to the delay parameter.
2. The method of claim 1, wherein the method of obtaining an array of values and an array of symbols further comprises:
and judging whether the maximum value of the convolution absolute value in the window is smaller than a preset test threshold value or not, if so, discarding the maximum value of the absolute value and not storing the maximum value into a numerical value array and a symbol array.
3. The method of claim 2, wherein said obtaining the relative delay time for the speaker to arrive at the receiving location comprises:
and acquiring the maximum value in the numerical array, and calculating according to the position of the maximum value in the numerical array to obtain the relative delay of the loudspeaker to be tested.
4. The method of claim 3, wherein the obtaining the relative delay time for the speaker to arrive at the receiving location further comprises:
acquiring the sign corresponding to the maximum numerical value in a symbol array;
and obtaining the wiring method of the phase of the loudspeaker to be tested according to the sign corresponding to the maximum value.
5. The utility model provides a speaker time delay automatic adjustment's equipment, its characterized in that includes the speaker and the test equipment of awaiting measuring, wherein, the test equipment includes:
the receiving unit is used for receiving the audio signal which is sent by any loudspeaker to be tested and modulated by the test pulse signal;
the relative delay calculating unit is used for carrying out convolution operation on the received audio signal and the test pulse signal at a receiving position by taking the time of sending the audio signal by the loudspeaker as the initial test time to obtain the relative delay of the loudspeaker reaching the receiving position; the test pulse signals are selected according to the pronunciation characteristics of the loudspeaker;
the relative delay calculating unit comprises a window sliding subunit, a time delay calculating subunit and a time delay calculating unit, wherein the window sliding subunit is used for acquiring an audio signal segment from the audio signal received within the test limit time in a sliding window mode, the time length of the sliding window is consistent with the time length of the test pulse signal, and the stepping length of the sliding window is the length of an integer number of data points within the window;
the array generating subunit is used for convolving the audio signal segment in each window in the sliding process with the test pulse signal, acquiring the maximum value of the convolution absolute value in the window, and respectively storing the value of the maximum value of the absolute value and the sign corresponding to the value into an initially empty value array and a sign array;
the delay parameter calculating unit is used for calculating the delay parameter corresponding to each loudspeaker to be detected according to the relative delay corresponding to all the loudspeakers in the room to be detected respectively;
and the output control unit is used for carrying out delay adjustment on the loudspeaker corresponding to the delay parameter according to the delay parameter.
6. The apparatus of claim 5, wherein the array generation subunit further comprises:
and the threshold judgment subunit is used for judging whether the maximum value of the convolution absolute value in the window is smaller than a preset test threshold, and if the maximum value of the convolution absolute value in the window is smaller than the preset test threshold, the maximum value of the absolute value is discarded and is not stored in the numerical value array and the symbol array.
7. The apparatus of claim 6, wherein the relative delay calculating unit further comprises:
and the relative delay obtaining subunit is used for calculating the relative delay of the loudspeaker to be tested according to the position of the maximum numerical value in the numerical value array.
8. The apparatus of claim 7, wherein the relative delay calculating unit further comprises:
and the phase judgment subunit is used for acquiring the wiring direction of the phase of the loudspeaker to be tested according to the sign corresponding to the maximum value.
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 CN109862503A (en) | 2019-06-07 |
| CN109862503B true 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) |
Families Citing this family (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 |
Family Cites Families (17)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP4422941B2 (en) * | 2002-01-18 | 2010-03-03 | 株式会社ケンウッド | Sound field adjustment device |
| DK1981310T3 (en) * | 2007-04-11 | 2017-09-18 | Oticon As | Hearing aid with linearized output stage |
| CN101321201B (en) * | 2007-06-06 | 2011-03-16 | 联芯科技有限公司 | Echo elimination device, communication terminal and method for confirming echo delay time |
| CN101448186B (en) * | 2007-11-26 | 2012-07-18 | 鸿富锦精密工业（深圳）有限公司 | System and method for automatic regulating sound effect of a loudspeaker |
| TWI403188B (en) * | 2007-12-07 | 2013-07-21 | Hon Hai Prec Ind Co Ltd | System and method for automatic adjusting sound of speakers |
| CN101938686B (en) * | 2010-06-24 | 2013-08-21 | 中国科学院声学研究所 | Measurement system and measurement method for head-related transfer function in common environment |
| WO2013016500A1 (en) * | 2011-07-28 | 2013-01-31 | Thomson Licensing | Audio calibration system and method |
| EP2949133B1 (en) * | 2013-01-24 | 2019-02-13 | Dolby Laboratories Licensing Corporation | Automatic loudspeaker polarity detection |
| CN103414980B (en) * | 2013-07-30 | 2016-09-07 | 浙江中科电声研发中心 | The time delay of a kind of multi output channel sound systems and the compensation method of Frequency Response |
| GB2535990A (en) * | 2015-02-26 | 2016-09-07 | Univ Antwerpen | Computer program and method of determining a personalized head-related transfer function and interaural time difference function |
| CN106331951B (en) * | 2016-08-18 | 2019-10-11 | 苏州上声电子股份有限公司 | A kind of harmonic distortion measurement method and system based on continuous logarithmic frequency sweep |
| EP3300389B1 (en) * | 2016-09-26 | 2021-06-16 | STMicroelectronics (Research & Development) Limited | A speaker system and method |
| CN106686520B (en) * | 2017-01-03 | 2019-04-02 | 南京地平线机器人技术有限公司 | The multi-channel audio system of user and the equipment including it can be tracked |
| CN109215667B (en) * | 2017-06-29 | 2020-12-22 | 华为技术有限公司 | Time delay estimation method and device |
| CN107785026B (en) * | 2017-10-18 | 2020-10-20 | 会听声学科技（北京）有限公司 | Time delay estimation method for indoor echo cancellation of set top box |
| CN109246570B (en) * | 2018-08-29 | 2020-12-11 | 北京声智科技有限公司 | Microphone quality inspection device and method |
| CN109102821B (en) * | 2018-09-10 | 2021-05-25 | 思必驰科技股份有限公司 | Time delay estimation method, time delay estimation system, storage medium and electronic equipment |
-
2019
- 2019-01-30 CN CN201910089278.3A patent/CN109862503B/en active Active
Also Published As
| Publication number | Publication date |
|---|---|
| CN109862503A (en) | 2019-06-07 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| CN109862503B (en) | Method and equipment for automatically adjusting loudspeaker delay | |
| US8175284B2 (en) | Method and apparatus for calibrating sound-reproducing equipment | |
| US8798280B2 (en) | Calibration method and device in an audio system | |
| JP6572894B2 (en) | Information processing apparatus, information processing method, and program | |
| CN105301594B (en) | Range measurement | |
| Ratnam et al. | Blind estimation of reverberation time | |
| CN109275084A (en) | Test method, device, system, equipment and the storage medium of microphone array | |
| US8996367B2 (en) | Sound processing apparatus, sound processing method and program | |
| US7492908B2 (en) | Sound localization system based on analysis of the sound field | |
| US8818805B2 (en) | Sound processing apparatus, sound processing method and program | |
| RU2635286C2 (en) | Method and device for determining microphone position | |
| CN107360530B (en) | Echo cancellation testing method and device | |
| EP2203002B1 (en) | Method for measuring frequency characteristic and rising edge of impulse response, and sound field correcting apparatus | |
| CN111182435B (en) | Testing method and device of voice equipment | |
| CN107645696B (en) | One kind is uttered long and high-pitched sounds detection method and device | |
| CN108882115B (en) | Loudness adjustment method and device and terminal | |
| US9560461B2 (en) | Automatic loudspeaker polarity detection | |
| WO2005104610A1 (en) | Resonance frequency determining method, resonance frequency selecting method, and resonance frequency determining apparatus | |
| WO2020163419A1 (en) | Intelligent personal assistant | |
| JP5627440B2 (en) | Acoustic apparatus, control method therefor, and program | |
| US20170289727A1 (en) | Sound field control system, analysis device, and accoustic device | |
| WO2023051622A1 (en) | Method for improving far-field speech interaction performance, and far-field speech interaction system | |
| CN112995882B (en) | Intelligent equipment audio open loop test method |
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 |