CN106817655B - Speaker control method and device - Google Patents

Speaker control method and device Download PDF

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Publication number
CN106817655B
CN106817655B CN201510868769.XA CN201510868769A CN106817655B CN 106817655 B CN106817655 B CN 106817655B CN 201510868769 A CN201510868769 A CN 201510868769A CN 106817655 B CN106817655 B CN 106817655B
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signal
loudspeaker
current time
calculated
gain coefficient
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CN106817655A (en
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蒋斌
纪伟
吴晟
林福辉
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Spreadtrum Communications Shanghai Co Ltd
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Spreadtrum Communications Shanghai Co Ltd
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Abstract

Speaker control method and device, which comprises obtain the digital signal of input;The digital signal of input is filtered and buffered respectively, obtains the buffering signals at current time;Based on the voltage signal and current signal at loudspeaker both ends described in current time, the gain of the buffering signals is adjusted, within a preset range by the structural behaviour state modulator of the loudspeaker.The quality of the audio signal of loudspeaker output can be improved in above-mentioned scheme.

Description

Speaker control method and device
Technical field
The present invention relates to fields of communication technology, more particularly to a kind of speaker control method and device.
Background technique
Loudspeaker is a kind of energy transducer for electric signal being changed into acoustical signal, and the performance superiority and inferiority of loudspeaker is to sound quality It influences very big.Under normal circumstances, when the power of input signal is larger, loudspeaker can generate non-linear distortion or lead to voice coil With the damage of vibrating diaphragm.The prior art is generally by way of reducing input signal power, to avoid the generation of above situation.
But this speaker control method, the power of speaker output signal is always worked at into rated power hereinafter, Since rated power is smaller, and most resonant frequency audio signal below is filtered out, so that the audio of loudspeaker output Signal it is second-rate.
Summary of the invention
The technical issues of embodiment of the present invention solves is the quality for improving the audio signal of loudspeaker output.
To solve the above problems, the embodiment of the invention provides a kind of speaker control methods, which comprises
Obtain the digital signal of input;
The digital signal of input is filtered and buffered respectively, obtains the buffering signals at current time;
Based on the voltage signal and current signal at loudspeaker both ends described in current time, to the gains of the buffering signals into Row is adjusted, within a preset range by the structural behaviour state modulator of the loudspeaker.
Optionally, the structural behaviour parameter of the loudspeaker includes at least one of voice coil temperature and vibrating diaphragm displacement.
Optionally, the voltage signal and current signal based on loudspeaker both ends described in current time, to the buffering The gain of signal is adjusted, comprising:
Down-sampled processing is carried out to acquired real-time voltage signal and current signal respectively;
Based on the voltage signal and current signal after down-sampled, calculating separately to obtain current time controls voice coil temperature The temperature gain coefficient of system and the displacement gain coefficient that vibrating diaphragm displacement is controlled;
The maximum permissible voltage of the power amplifier of voltage and the loudspeaker based on the buffering signals, is calculated The signal gain coefficient that the voltage of the buffering signals is controlled;
Based on the signal gain coefficient, temperature gain coefficient and displacement gain coefficient, the buffering signals are calculated Final gain coefficient;
Gain process is carried out to the buffering signals using the final gain coefficient.
Optionally, the temperature control gain is calculated in the voltage signal and current signal based on after down-sampled Coefficient, comprising:
According to the voltage signal and current signal after down-sampled, the resistance of voice coil direct current is calculated;
Using the voice coil direct current resistance being calculated, the voice coil temperature at current time is calculated;
The maximum allowable temperature of voice coil temperature and the voice coil based on current time, is calculated the temperature gain system Number.
Optionally, the voltage signal and current signal based on after down-sampled, the displacement that current time is calculated increase Beneficial coefficient, comprising:
Using the voltage signal and current signal after down-sampled, the impedance transmitting letter of the loudspeaker at current time is calculated Number;
According to the impedance transfer function, the electric voltage displacement transmission function of the loudspeaker is determined;
According to the electric voltage displacement transmission function of the loudspeaker and it is down-sampled after buffering signals, current time is calculated Vibrating diaphragm displacement;
The maximum allowable displacement of vibrating diaphragm displacement and the vibrating diaphragm based on current time, is calculated the displacement gain system Number.
Optionally, when the input signal to current time is filtered, trapper in used filter group Centre frequency be the loudspeaker current time resonant frequency.
Optionally, the resonant frequency at the loudspeaker current time is calculated using following formula:
Wherein, f0(n) resonant frequency at the n-th moment of loudspeaker, Z (n, f) are indicated For the impedance transfer function of frequency domain, max () indicates maximum value, and abs () indicates the amplitude of plural number.
Optionally, it is described it is down-sampled after voltage signal and the sample rate of current signal meet following formula:
4·f0≤fs_ctrl≤10·f0, wherein fs_ctrlThe sampling of voltage signal and current signal after indicating down-sampled Rate, f0Indicate the loudspeaker in the resonant frequency at current time.
Optionally, the final gain coefficient of the buffering signals is calculated using following formula:
gtot(n)=min (gs(n),gx(n))*gT(n), wherein gtot(n) buffering signals described in current time are indicated most Whole gain coefficient, gs(n) the signal gain coefficient, g are indicatedT(n) the temperature gain coefficient, g are indicatedx(n) institute's rheme is indicated Move gain coefficient.
The embodiment of the invention also provides a kind of loudspeaker controller, described device includes:
Acquiring unit, suitable for obtaining the digital signal of input;
Filter processing unit is filtered respectively suitable for the digital signal to input;
Buffered unit obtains buffering signals suitable for carrying out buffered to the signal after filtering processing;
Gain processing unit, suitable for voltage signal and current signal based on loudspeaker both ends described in current time, to institute The gain for stating buffering signals is adjusted, within a preset range by the structural behaviour state modulator of the loudspeaker.
Optionally, the structural behaviour parameter of the loudspeaker includes at least one of voice coil temperature and vibrating diaphragm displacement.
Optionally, the gain processing unit includes:
Down-sampled subelement, suitable for carrying out down-sampled processing to acquired real-time voltage signal and current signal respectively;
Temperature controls subelement, suitable for current time is calculated based on the voltage signal and current signal after down-sampled The temperature gain coefficient that voice coil temperature is controlled;
Bit andits control subelement, suitable for current time is calculated based on the voltage signal and current signal after down-sampled The displacement gain coefficient controlled is displaced to vibrating diaphragm;
Signal controls subelement, and the power amplifier suitable for voltage and the loudspeaker based on the buffering signals is most It is big to allow voltage, the signal gain coefficient controlled the voltage of the buffering signals is calculated;
Computation subunit is suitable for being based on the signal gain coefficient, temperature gain coefficient and displacement gain coefficient, calculate To the final gain coefficient of the buffering signals;
Gain process subelement is suitable for carrying out gain process to the buffering signals using the final gain coefficient.
Optionally, temperature control subelement is suitable for being calculated according to the voltage signal and current signal after down-sampled It is hindered to voice coil direct current;Using the voice coil direct current resistance being calculated, the voice coil temperature at current time is calculated;Sound based on current time The maximum allowable temperature for enclosing temperature and the voice coil, is calculated the temperature gain coefficient.
Optionally, voltage signal and current signal of the Bit andits control subelement suitable for use after down-sampled, calculates To the impedance transfer function of the loudspeaker at current time;According to the impedance transfer function, the voltage position of the loudspeaker is determined Move transmission function;According to the electric voltage displacement transmission function of the loudspeaker and it is down-sampled after buffering signals, be calculated current The vibrating diaphragm at moment is displaced;The maximum allowable displacement of vibrating diaphragm displacement and the vibrating diaphragm based on current time, is calculated institute's rheme Move gain coefficient.
Optionally, the filter processing unit is used when the input signal to current time is filtered The centre frequency of trapper is the resonant frequency at the loudspeaker current time in filter group.
Optionally, the resonant frequency at the loudspeaker current time is calculated using following formula:
Wherein, f0(n) resonant frequency at the n-th moment of loudspeaker, Z (n, f) are indicated For the impedance transfer function of frequency domain, max () indicates maximum value, and abs () indicates the amplitude of plural number.
Optionally, it is described it is down-sampled after voltage signal and the sample rate of current signal meet following formula:
4·f0≤fs_ctrl≤10·f0, wherein fs_ctrlThe sampling of voltage signal and current signal after indicating down-sampled Rate, f0Indicate the loudspeaker in the resonant frequency at current time.
Optionally, the computation subunit is suitable for calculating the gain coefficient at current time using following formula:
gtot(n)=min (gs(n),gx(n))*gT(n), wherein gtot(n) increasing of buffering signals described in current time is indicated Beneficial coefficient, gs(n) the signal gain coefficient, g are indicatedT(n) the temperature gain coefficient, g are indicatedx(n) indicate that the displacement increases Beneficial coefficient.
Compared with prior art, technical solution of the present invention has the advantage that
Above-mentioned scheme, the voltage signal and current signal fed back by loudspeaker both ends believe the input at current time Number gain controlled, can input signal power be higher than loudspeaker rated power when, it is ensured that loudspeaker safety Work, and the output quality of audio signal can be promoted, promote the usage experience of user.
Further, the voltage signal and current signal fed back by loudspeaker both ends, can voice coil temperature to loudspeaker In default range, to avoid the damage of loudspeaker, the safety of speaker operation can be improved in degree and vibrating diaphragm Bit andits control.
Further, it is filtered to the audio signal that current time inputs, the centre frequency of trapper is set The frequency that loudspeaker is set in the resonant frequency at current time, near influence and resonant frequency of the adjustable resonant frequency to displacement Response, and then the audio signal quality of output can be improved.
Further, when being sampled to down-sampled treated voltage signal and current signal, by sample frequency control System can reduce calculation amount between 4 times and 10 times of the resonant frequency at the loudspeaker current time, save computing resource.
Detailed description of the invention
Fig. 1 is the flow chart of one of embodiment of the present invention speaker control method;
Fig. 2 is the flow chart of another speaker control method in the embodiment of the present invention;
Fig. 3 is a kind of method flow for calculating signal gain coefficient of the loudspeaker at current time in the embodiment of the present invention Figure;
Fig. 4 is a kind of method flow for calculating displacement gain coefficient of the loudspeaker at current time in the embodiment of the present invention Figure;
Fig. 5 is a kind of method flow for calculating temperature gain coefficient of the loudspeaker at current time in the embodiment of the present invention Figure;
Fig. 6 is the structural schematic diagram of the loudspeaker controller in the embodiment of the present invention;
Fig. 7 is the corresponding operation principle schematic diagram of loudspeaker controller described in Fig. 6.
Specific embodiment
To solve the above-mentioned problems in the prior art, technical solution used in the embodiment of the present invention passes through loudspeaker two The voltage signal and current signal for holding feedback, control the gain of the input signal at current time, can be in input signal Power when being higher than the rated power of loudspeaker, it is ensured that loudspeaker trouble free service, and the output matter of audio signal can be promoted Amount, promotes the usage experience of user.
To make the above purposes, features and advantages of the invention more obvious and understandable, with reference to the accompanying drawing to the present invention Specific embodiment be described in detail.
Fig. 1 shows the flow chart of one of embodiment of the present invention speaker control method.Loudspeaker as shown in Figure 1 Control method may include:
Step S101: the digital signal of input is obtained.
Step S102: being filtered the digital signal of input respectively and buffered, obtains the buffering letter at current time Number.
Step S103: voltage signal and current signal based on loudspeaker both ends described in current time believe the buffering Number gain be adjusted, within a preset range by the structural behaviour state modulator of the loudspeaker.
In specific implementation, the structural behaviour parameter of the loudspeaker includes at least one of voice coil temperature, vibrating diaphragm displacement.
Above-mentioned scheme, the voltage signal and current signal fed back by loudspeaker both ends believe the input at current time Number gain controlled, can input signal power be higher than loudspeaker rated power when, it is ensured that loudspeaker safety Work, and the output quality of audio signal can be promoted, promote the usage experience of user.
Further details of introduction is made to one of embodiment of the present invention speaker control method below in conjunction with Fig. 2.
Step S201: the input signal at current time is obtained.
In specific implementation, it is assumed that the digital signal of input is divided into the muti-piece number that every block length is k (k >=1) and believes Number, the digital signal at current time, i.e., n-th piece of digital signal can be expressed as s (n).
Step S202: the input signal at current time is filtered.
In specific implementation, when getting the input signal at current time, can input signal to current time into Row filtering processing, to eliminate interference caused by specific frequency.Wherein, it is filtered in the input signal to current time When, it can be filtered using the filter group including multiple cascading filters, used filter can be limited Impact response filter (FIR), or IIR filter (IIR).
In specific implementation, the transmission function of the filter group is expressed as hfb(n), then, by filtering processing after Obtained signal sfb(n) it can indicate are as follows:
Wherein,For convolution symbol.
In specific implementation, each filter in multiple cascading filters, such as low-pass filter, high-pass filter, band logical The centre frequency of filter, trapper etc. can be preset numerical value, can also be configured according to the actual needs.
For example, in an embodiment of the present invention, in order to adjust near influence and resonant frequency of the resonant frequency to displacement Frequency response can set the centre frequency of trapper to loudspeaker in the resonant frequency f at current time0(n)。
In specific implementation, resonant frequency f of the loudspeaker at current time0It (n) is changed with the time, it can be with Voltage signal and current signal by loudspeaker at current time both ends are estimated to obtain.
Specifically, down-sampled obtain can be carried out for the voltage signal at current time loudspeaker both ends and current signal first To the current signal i after down-sampledd(n) and voltage signal ud(n)。
Then, the impedance transfer function Z (n) of loudspeaker can be calculated using following formula:
Above-mentioned formula (2), which is converted into frequency domain, to be indicated are as follows:
Id(n, f) Z (n, f)=Ud(n,f) (3)
Wherein, Id(n, f) indicate frequency domain it is down-sampled after current signal, Ud(n, f) indicate frequency domain it is down-sampled after electricity Signal is pressed, Z (n, f) indicates the impedance transfer function of frequency domain.
After obtaining the impedance transfer function Z (n, f) of corresponding frequency domain, the resonant frequency at the loudspeaker current time f0(n) it is calculated using following formula:
Wherein, f0(n) indicate that the resonant frequency at the n-th moment of loudspeaker, Z (n, f) are the impedance transfer function of frequency domain, max () expression is maximized, and abs () indicates the amplitude of plural number.
Step S203: buffered is carried out to the input signal after filtering processing, obtains buffering signals.
In specific implementation, the input signal after filtering processing is buffered, obtained buffering signals can indicate are as follows:
sla(n)={ sfb(n-t),sfb(n-t+1),…,sfb(n)} (5)
Wherein, sla(n) indicate that the buffering signals, t indicate that buffer time is the corresponding time span of t block digital signal, And t >=0.
Step S204: the buffering signals are calculated in the signal gain coefficient at current time.
In specific implementation, it can control the voltage of the buffering signals most no more than the power amplifier in loudspeaker It is big to allow voltage, the signal gain coefficient that the voltage to determine to buffering signals is controlled.Specifically, referring to Fig. 3 institute Show, may include following step S301~S305:
Step S301: being smoothed the buffering signals, and calculates the amplitude of the buffering signals after smoothing processing Maximum value.
It in specific implementation, can be first to buffering signals sla(n) it is smoothed, and is calculated using following formula Obtain smoothed out buffering signals sla(n) maximum value of amplitude:
Sa(n)=(1- α) * Sa(n-1)+α*max(abs(sla(n))) (6)
Wherein, SaIt (n) is the maximum value of the n-th block signal (the corresponding time is the current time) amplitude, Sa(n-1) it is The maximum value of (n-1)th block signal amplitude, α indicates preset coefficient, and 0≤α≤1, abs () indicate signed magnitude arithmetic(al), max () Expression is maximized operation.
Step S302: the gain of the buffering signals after calculating smoothing processing.
In specific implementation, the buffering signals s after smoothing processing can be calculated using following formulala(n) gain:
Wherein, gsa(n) be the n-th block signal gain, thrd1 is preset gain threshold, gmaxIt is that preset gain is maximum Value.
Step S303: amplitude limiting processing is carried out to the buffering signals after smoothing processing, the buffering signals after obtaining amplitude limiting processing.
In specific implementation, the maximum value of the signal after clipping can be calculated using following formula:
Sl(n)=(1- β) * Sl(n-1)+β*max(abs(sla(n)))·gsa(n) (8)
Wherein, SlIt (n) is n-th piece of limitation signal Amplitude maxima, β indicates preset coefficient and 0≤β≤1.
Step S304: the gain of the buffering signals after calculating amplitude limiting processing.
In specific implementation, the gain of limitation signal can be calculated using following formula:
Wherein, gsl(n) be n-th piece of limitation signal gain, thrd2 is preset limiting threshold.
Step S305: it according to the gain of the buffering signals after the gain of the buffering signals after smoothing processing and clipping, calculates Obtain the signal gain coefficient of the buffering signals.
In specific implementation, the signal gain coefficient can be calculated using following formula:
gs(n)=gsa(n)*gsl(n) (10)
Step S205: the buffering signals are calculated in the displacement gain coefficient at current time.
In specific implementation, the maximum allowable of voice coil can be no more than in the voice coil temperature at current time by control voice coil Temperature, to determine the displacement gain coefficient of the buffering signals.Specifically, it is shown in Figure 4, it may include following step Rapid S401~S404:
Step S401: down-sampled processing is carried out to acquired voltage signal and current signal respectively.
In specific implementation, due to when estimating voice coil temperature, the frequency of temperature measurement at preset low frequency, It, can be first to current time and to vibrating diaphragm displacement contribute that maximum frequency range is the resonant frequency of loudspeaker 4 times hereinafter, therefore The voltage signal and current signal at loudspeaker both ends carry out down-sampled processing.
In specific implementation, the voice coil temperature at current time in order to obtain, can be to the voltage signal and electricity after down-sampled Stream signal is sampled.Wherein, in order to cover vibrating diaphragm displacement where main region, to it is described it is down-sampled after voltage signal ud(n) and current signal id(n) sample frequency f when being sampleds_ctrlIt can satisfy:
4·f0≤fs_ctrl≤10·f0 (11)
Wherein, f0Indicate loudspeaker in the resonant frequency at current time.
It is to be herein pointed out resonant frequency f of the loudspeaker at current time0It can be preset numerical value, it can also be with For the resonant frequency f at the loudspeaker current time0(n), resonant frequency f in step S203 is specifically referred to0(n) introduction.
Step S402: loudspeaker described in current time is calculated in voltage signal and current signal after use is down-sampled Impedance transfer function.
In specific implementation, obtain it is down-sampled after current signal id(n) and voltage signal ud(n) it after, can use The impedance transfer function Z (n) of loudspeaker is calculated in following formula:
Above-mentioned formula (12), which is converted into frequency domain, to be indicated are as follows:
Id(n, f) Z (n, f)=Ud(n,f) (13)
Wherein, Id(n, f) indicate frequency domain it is down-sampled after current signal id(n), Ud(n, f) indicates the down-sampled of frequency domain Voltage signal u afterwardsd(n), Z (n, f) indicates the impedance transfer function Z (n) of frequency domain.
Step S403: according to the impedance transfer function of the loudspeaker at the current time, current time loudspeaker is determined Electric voltage displacement transmission function.
It in specific implementation, can be using such as after obtaining impedance transfer function Z (n) of the loudspeaker at current time Under formula obtain the electric voltage displacement transmission function h in current time loudspeakervx(n):
Wherein, Hvx(n, f) is the frequency domain representation of electric voltage displacement transmission function, Ze(n, f) is the frequency domain representation of voice coil impedance, Bl is the electrical power conversion factor of loudspeaker.
Step S404: according to the electric voltage displacement transmission function of the loudspeaker at current time and it is down-sampled after buffering signals, The vibrating diaphragm displacement of loudspeaker described in current time is calculated.
In specific implementation, in the electric voltage displacement transmission function for the loudspeaker for determining current time, first can will before State the buffering signals s that step S204 is obtainedla(n) carry out it is down-sampled processing obtain it is down-sampled after buffering signals sd(n).Then, Can based on obtain it is down-sampled after buffering signals sd(n) and electric voltage displacement transmission function hvx(n), it is calculated using following formula Loudspeaker is obtained to be displaced in the vibrating diaphragm at current time:
Wherein, xla(n) indicate that loudspeaker is displaced in the vibrating diaphragm at current time, sd(n) buffering signals after indicating down-sampled, hvx(n) electric voltage displacement transmission function is indicated.
Above-mentioned formula (15), which is converted to frequency domain, to be indicated are as follows:
Xla(n, f)=Sd(n,f)·Hvx(n,f) (16)
Wherein, Xla(n, f) is xla(n) indicate that the loudspeaker of frequency domain is displaced in the vibrating diaphragm at current time, Sd(n, f) indicates frequency Domain it is down-sampled after buffering signals sd(n), HvxThe electric voltage displacement transmission function h of (n, f) expression frequency domainvx(n)。
Step S206: the buffering signals are calculated in the temperature gain coefficient at current time.
In specific implementation, the voice coil can be no more than most in the voice coil temperature at current time by control loudspeaker It is big to allow temperature value, to determine the temperature gain coefficient of the buffering signals.Specifically may refer to shown in Fig. 5, may include as Under step S501~S504:
Step S501: the voltage signal to current time loudspeaker both ends and current signal carry out down-sampled processing respectively.
In specific implementation, the introduction of this step refers to step S401 in Fig. 4, and details are not described herein.
Step S502: according to the voltage signal and current signal after down-sampled, the resistance of voice coil direct current is calculated.
In specific implementation, voice coil direct current hinders ReThe resistance of voice coil direct current can be by the current signal i after down-sampledd(n) and voltage Signal ud(n) it is calculated:
Re=ud(n,fpt)/id(n,fpt) (17)
Wherein, ud(n,fpt) and id(n,fpt) indicate in fptThe range value of neighbouring signal.
Step S503: using the voice coil direct current resistance being calculated, the voice coil temperature at current time is calculated.
In specific implementation, voice coil direct current hinders ReIt is changed linearly with temperature, is obtaining voice coil mainstream resistance ReIt later, can be with The voice coil temperature at current time is calculated using following formula:
T=T0+α·(Re-R0)/R0 (18)
Wherein, T indicates the voice coil temperature at current time, and α indicates preset temperature-coefficient of electrical resistance, T0Indicate voice coil initial The temperature at moment, R0Indicate the resistance value that voice coil is carved at the beginning.
Step S504: it according to the maximum allowable temperature of the voice coil temperature at current time and the voice coil, is calculated described The temperature gain coefficient of buffering signals.
In specific implementation, be referred to how the voltage based on the buffering signals be no more than power amplifier maximum Voltage determines the mode of the signal gain coefficient, according to the maximum allowable temperature of the voice coil temperature at current time and the voice coil The temperature gain coefficient of the buffering signals is calculated in degree, and this will not be repeated here by the present invention.
It is to be herein pointed out above-mentioned step S204~S206 execution sequence and be not particularly limited, ability The technical staff in domain can be configured according to the actual needs, and the present invention is herein with no restrictions.
Step S207: signal gain coefficient, temperature gain coefficient and displacement gain coefficient based on the buffering signals, meter Calculation obtains the buffering signals in the final gain coefficient at current time.
It in the present invention one is implemented, can be based on that temperature gain coefficient, displacement gain is calculated using following formula Coefficient and signal gain coefficient calculate the gain coefficient of buffering signals described in current time:
gtot(n)=min (gs(n),gx(n))*gT(n) (19)
Wherein, gtot(n) the final gain coefficient of the buffering signals at current time, and g are indicateds(n) >=0, gs(n) institute is indicated State signal gain coefficient and gs(n) >=0, gT(n) the temperature gain coefficient and 0≤g are indicatedT(n)≤1, gx(n) described in indicating Displacement gain coefficient and gx(n)≥0。
It is of course also possible to use other modes obtain based on be calculated temperature gain coefficient, displacement gain coefficient and Signal gain coefficient calculates the final gain coefficient of buffering signals described in current time, and the present invention is herein with no restrictions.
Step S208: using the final gain coefficient being calculated, carrying out adaptive gain processing to the buffering signals, Signal after obtaining gain process.
In specific implementation, as the buffering signals s that current time is calculatedla(n) gain coefficient gtot(n) it Afterwards, it can use and the buffering signals s is calculatedla(n) gain coefficient gtot(n) to buffering signals sla(n) it carries out adaptive Gain process is answered, the signal after finally obtained gain process can indicate are as follows:
sag(n)=sfb(n-t)*gtot(n) (20)
Wherein, sag(n) signal after gain process, s are indicatedfb(n-t) the signal buffer signal of the n-th-t block is indicated.
Step S209: the signal after gain process is subjected to digital-to-analogue conversion, obtains analog signal.
In specific implementation, in order to estimate voice coil temperature, can signal after determining gain process in low frequency When sinusoidal signal is added, as temperature measurement signal.Wherein, preset sinusoidal signal, which is added, to be indicated are as follows:
Wherein, spt(n) sinusoidal signal being added at default low frequency, f are indicatedptFor temperature measurement signal spt(n) frequency Rate, fsFor the signal s after gain processag(n) sample frequency;A is temperature measurement signal spt(n) amplitude, and far smaller than Audio frequency signal amplitude when loudspeaker works normally, i indicate the ith sample point in n-th piece of digital signal corresponding time, k Indicate the length of every piece of digital signal.
It is to be herein pointed out the preset temperature measurement signal frequency fptNumerical value can be according to the actual needs It is configured.For example, in an embodiment of the present invention, fptIt can be set to 40Hz.In addition, in order to protect temperature measurement signal Validity, can be in fptIt is nearby filtered, filters out the interference of other neighbouring signals.
In specific implementation, the signal obtained after temperature measurement signal, which is added, to be expressed as follows:
stot(n)=sag(n)+spt(n) (22)
Wherein, stot(n) signal after gain process, s are indicatedag(n) temperature measurement signal is added at preset low frequency spt(n) signal obtained after.
In specific implementation, signal s is being obtainedtotIt (n), can be to signal s aftertot(n) digital-to-analogue conversion is carried out, it will be digital Signal is converted to corresponding analog signal output.
Step S210: the loudspeaker is input to after the analog signal input that conversion obtains is amplified processing.
In specific implementation, the analog signal obtained after digital-to-analogue conversion is sent into after power amplifier amplifies processing and is inputted To loudspeaker, and export to user.
It is former below in conjunction with the corresponding device of speaker control method of the Fig. 6 to Fig. 7 to the embodiment of the present invention and its work Reason is further described in detail.
Fig. 6 shows the structural schematic diagram of one of embodiment of the present invention loudspeaker controller.It is as shown in FIG. 6 to raise Sound device control device 600 may include acquiring unit 601, filter processing unit 602, buffered unit 603 and gain process Unit 604, in which:
The acquiring unit 601, suitable for obtaining the digital signal of input.
The filter processing unit 602, is filtered respectively suitable for the digital signal to input.
In specific implementation, the filter processing unit 602 is when the input signal to current time is filtered, The centre frequency of trapper is the resonant frequency at the loudspeaker current time in used filter group.
In specific implementation, the resonant frequency at the loudspeaker current time is calculated using following formula:
Wherein, f0(n) resonant frequency at the n-th moment of loudspeaker, Z (n, f) are indicated For the impedance transfer function of frequency domain, max () indicates maximum value, and abs () indicates the amplitude of plural number.
In specific implementation, it is described it is down-sampled after voltage signal and the sample rate of current signal meet following formula:
4·f0≤fs_ctrl≤10·f0, wherein fs_ctrlThe sampling of voltage signal and current signal after indicating down-sampled Rate, f0Indicate the loudspeaker in the resonant frequency at current time.
The buffered unit 603 obtains buffering signals suitable for carrying out buffered to the signal after filtering processing.
The gain processing unit 604, suitable for voltage signal and the electric current letter based on loudspeaker both ends described in current time Number, the gain of the buffering signals is adjusted, within a preset range by the structural behaviour state modulator of the loudspeaker.
In specific implementation, the structural behaviour parameter of the loudspeaker includes at least one in voice coil temperature and vibrating diaphragm displacement .
In specific implementation, the gain processing unit 604 may include down-sampled subelement 6041, temperature control son list Member 6042, Bit andits control subelement 6043, signal control subelement 6044, computation subunit 6045 and gain process subelement 6046, in which:
The down-sampled subelement 6041, suitable for carrying out down-sampled place to acquired real-time voltage and current signal respectively Reason.
The temperature controls subelement 6042, suitable for being calculated based on the voltage signal and current signal after down-sampled The temperature gain coefficient that current time controls voice coil temperature.
In specific implementation, temperature control subelement 6042 be suitable for according to after down-sampled voltage signal and electric current believe Number, the resistance of voice coil direct current is calculated;Using the voice coil direct current resistance being calculated, the voice coil temperature at current time is calculated;Based on working as The temperature gain coefficient is calculated in the maximum allowable temperature of the voice coil temperature at preceding moment and the voice coil.
The Bit andits control subelement 6043, suitable for being calculated based on the voltage signal and current signal after down-sampled Current time is displaced the displacement gain coefficient controlled to vibrating diaphragm.
In specific implementation, voltage signal and electric current letter of the Bit andits control subelement 6043 suitable for use after down-sampled Number, the impedance transfer function of the loudspeaker at current time is calculated;According to the impedance transfer function, the loudspeaker is determined Electric voltage displacement transmission function;According to the electric voltage displacement transmission function of the loudspeaker and it is down-sampled after buffering signals, calculate Obtain the vibrating diaphragm displacement at current time;The maximum allowable displacement of vibrating diaphragm displacement and the vibrating diaphragm based on current time, calculates To the displacement gain coefficient.
The signal controls subelement 6044, and the power suitable for voltage and the loudspeaker based on the buffering signals is put The maximum permissible voltage of big device, is calculated the signal gain coefficient controlled the voltage of the buffering signals.
The computation subunit 6045 is suitable for being based on the signal gain coefficient, temperature gain coefficient and displacement gain system Number, is calculated the final gain coefficient of the buffering signals.
In specific implementation, the computation subunit 6045 is suitable for calculating the gain system at current time using following formula Number:
gtot(n)=min (gs(n),gx(n))*gT(n), wherein gtot(n) buffering signals at current time are indicated Final gain coefficient, gs(n) the signal gain coefficient, g are indicatedT(n) the temperature gain coefficient, g are indicatedx(n) described in indicating Displacement gain coefficient.
The gain process subelement 6046 is suitable for carrying out gain to the buffering signals using the final gain coefficient Processing.
Fig. 6 and Fig. 7 will be recombined below to do in detail the working principle of the loudspeaker controller in the embodiment of the present invention Thin introduction.
It is shown in Figure 7, it is assumed that it is the digital signal of k, n-th piece of number letter that input signal, which is divided into every block length, Number it can be expressed as s (n), wherein k >=1.Then, digital signal s (n) input filter processing unit 602 is filtered place Reason.Wherein, when being filtered to digital signal s (n), used filter group may include multiple cascading filters, Such as low-pass filter, high-pass filter, bandpass filter, trapper various types filter.Multiple grades in filter group The centre frequency of connection filter can be preset numerical value, be also possible to loudspeaker in the centre frequency f at current time0.Wherein, Centre frequency f of the loudspeaker at current time0Can voltage signal based on loudspeaker both ends and current signal obtain, specifically ask Referring to subsequent introduction.
In specific implementation, filter type can be finite impulse response filter (FIR) or infinite impulse response The transmission function of entire filter group is expressed as h by filter (IIR)fb(n), filtered signal sfbIt (n) can be using public affairs Formula (1) indicates.
It then, can be to the signal s after filtering processingfb(n) input buffered unit 603 carries out buffered, it is assumed that Buffer time is t (t >=0) block digital signal, then, the signal s after bufferedfb(n), available buffering signals sla(n) it can be indicated with formula (5).
Then, by buffering signals sla(n) it is sent to gain processing unit 604 and carries out adaptive gain processing.Wherein, gain Processing unit 604 can control subelement 6042 by temperature, and Bit andits control subelement 6043 and signal control subelement 6044 In all or part of decision, ultimately generate current time, i.e. the gain g at the n-th momenttot(n), and using current time Gain gtot(n) to buffering signals sla(n) gain process, the signal s after obtaining gain process are carried outag(n)。
In an embodiment of the present invention, gain processing unit 604 can also include activity detection subelement 6046, with detection It whether include audio signal in input signal.When determining input signal is mute signal, activity detection subelement 6046 is exported Corresponding signal is to gain processing unit 604, so that gain processing unit 604 does not use adaptive gain to input signal Processing;Adaptive gain processing is carried out to input signal conversely, then controlling gain processing unit 604;And work as in input signal not It only include audio signal, when further including noise signal, then activity detection subelement 6046 can also control gain processing unit 604 Noise signal in input signal is inhibited, those skilled in the art can be configured according to the actual needs, this It invents without limitation.
In specific implementation, when in input signal including audio signal, gain processing unit 604 can be by below Mode carries out gain process to the buffering signals of input:
Signal control module 6044, can be to buffering signals sla(n) voltage is controlled, so that its final output voltage No more than the maximum voltage of power amplifier 6049, to obtain signal gain coefficient gs(n)。
Specifically, signal control module 6044 can be calculated using formula (8) to buffering signals s firstla(n) into The maximum value of the amplitude of the smoothed out signal of row.Then, buffering signals s is calculated using following formula (9)la(n) gain, can The maximum value of the signal after clipping to be calculated using formula (10), finally using formula (10) according to smoothing processing after The gain of buffering signals and the gain of the buffering signals after clipping, are calculated the signal gain coefficient of the buffering signals.
In specific implementation, temperature control subelement 6042 can control the voice coil temperature of loudspeaker, with loudspeaking Voice coil temperature of the device at current time is no more than the maximum allowable temperature of voice coil, to obtain the temperature gain coefficient of signal.
Specifically, the voice coil temperature at temperature control subelement 6042 current time in order to obtain, needs to determine first The voice coil direct current at current time hinders.In order to obtain current time voice coil direct current resistance, obtain effective temperature measurement signal, can To determine that the signal frequency that gain process subelement 6045 exports reaches at preset low frequency addition sinusoidal signal 701.Wherein, The sinusoidal signal being added can be indicated using formula (21).Wherein, in order to protect temperature measurement signal (be added sinusoidal signal) Validity, can be in the low frequency fptIt is nearby filtered, to filter out the interference of other neighbouring signals.
The signal s obtained after sinusoidal signal is added at preset low frequencytot(n) formula (22) can be expressed as.
It is to be herein pointed out temperature control subelement 6042 and Bit andits control subelement 6043 are calculated in loudspeaker It is the voltage signal by the loudspeaker of feedback at current time to the temperature gain coefficient and displacement gain coefficient at current time It is carried out on the basis of the loudspeaker real-time parameter estimated with current signal, therefore, in order to make it easy to understand, here can be first Detailed introduction first is done to the process for how carrying out loudspeaker real time parameter estimation 702.
Shown in Figure 7, when estimating loudspeaker implementation parameter, the voltage at acquisition loudspeaker both ends first is believed Number and current signal, and loudspeaker is subjected to digital-to-analogue conversion in current time voltage signal and current signal, is counted accordingly Word voltage signal and digital current signal, and loudspeaker parameters are carried out in real time based on digital voltage signal and digital current signal Estimation.
Specifically, when the structural behaviour parameter to loudspeaker carries out real-time estimation, due to the frequency of temperature measurement signal Rate contributes maximum frequency range usually in 4 times of resonant frequency f of loudspeaker at low frequency, and to vibrating diaphragm displacement0Below.Therefore, may be used To carry out down-sampled processing to digital voltage signal and digital current signal, and to down-sampled treated digital voltage signal and Digital current signal is sampled.Wherein, in order to cover the main region where vibrating diaphragm displacement, to the number electricity after down-sampled Press signal id(n) and digital current signal ud(n) when being sampled, used sample rate is fs_ctrlIt can satisfy formula (11)。
In specific implementation, obtain it is down-sampled after digital voltage signal id(n) and digital current signal ud(n) after, Loudspeaker can be calculated using formula (12) and obtain frequency domain in the impedance transfer function at current time, and using formula (13) Impedance transfer function.Loudspeaker is being obtained in the impedance transfer function Z (n) at current time or the impedance transfer function of frequency domain After Z (n, f), electric voltage displacement transmission function can be calculated using formula (14).
Meanwhile loudspeaker is being obtained in the impedance transfer function Z (n) at the current time or impedance transfer function Z of frequency domain After (n, f), loudspeaker can be calculated in the resonant frequency at current time using formula (4).
Here resonant frequency f of the loudspeaker being calculated at current time0(n) filter processing unit 602 can be given, with Set the centre frequency of trapper therein to the resonant frequency f at current time0(n).Meanwhile loudspeaker is at current time Resonant frequency f0(n) gain process subelement 6046 can be transferred to, with using formula (19) to the digital current after down-sampled The sample rate of signal and digital voltage signal is controlled.
In specific implementation, when obtain to it is down-sampled after digital voltage signal id(n) and digital current signal ud(n) it Afterwards, the voice coil direct current at current time can be calculated using formula (17) hinder Re
In specific implementation, voice coil direct current hinders ReIt is changed linearly with temperature, is obtaining voice coil direct current resistance ReIt later, can be with The voice coil temperature at current time is calculated using formula (18).
In specific implementation, temperature control subelement 6042 by control loudspeaker current time voice coil temperature T not More than the maximum allowable displacement T of loudspeakermax, temperature gain coefficient g can be obtainedT(n), specific process be referred to as The process what obtains signal gain coefficient executes to obtain, and details are not described herein by the present invention.
In specific implementation, Bit andits control subelement 6044 is before being calculated the displacement gain coefficient at current time, The buffering signals s that buffered unit 603 can be exporteda(n) carry out obtaining after down-sampled processing it is down-sampled after buffering Signal sd(n).Wherein, to buffering signals sa(n) down-sampled module used by down-sampled processing is carried out and to loudspeaker both ends The voltage signal and current signal of feedback carry out down-sampled subelement structure having the same used by down-sampled processing.
Obtain it is down-sampled after buffering signals sd(n) after, Bit andits control subelement 6044 can pass through formula (14) Electric voltage displacement transmission function h is calculated according to the impedance transfer function of loudspeakervx(n), and formula (15) basis can be used Buffering signals s after down-sampledd(n) and electric voltage displacement transmission function hvx(n) the vibrating diaphragm displacement x at current time is obtainedla(n)。
Finally, Bit andits control subelement 6044 can be by controlling loudspeaker in the vibrating diaphragm displacement x at current timela(n) not More than the maximum allowable displacement X of loudspeakermax, displacement gain coefficient g can be obtainedx(n), specific process be referred to as The process what obtains signal gain coefficient executes to obtain, and details are not described herein by the present invention.
In specific implementation, subelement is controlled when temperature controls subelement 6042, Bit andits control subelement 6043, signal After the signal gain coefficients of 6044 buffering signals exported respectively, temperature gain coefficient and displacement gain coefficient, it is single to calculate son Buffering signals s can be calculated using formula (19) in member 6045la(n) gain gtot(n), and it is transmitted to gain process son list Member 6046, the signal s to carry out gain process using to buffering signals, after obtaining gain processag(n)。
In specific implementation, in order to estimate voice coil temperature, can signal after determining gain process in low frequency When sinusoidal signal is added, to ensure the validity of temperature measurement signal.The sinusoidal signal .. of addition can using formula (21) into Row indicates, temperature measurement signal s is addedpt(n) signal stot(n) it can be indicated using formula (22).
The signal s that gain process subelement 6046 exportstot(n) it is transmitted in power amplifier 6048 and amplifies processing Afterwards, loudspeaker is transmitted to after being transferred to DAC6049 and carrying out digital-to-analogue conversion to be exported.
In an embodiment of the present invention, the voice coil direct current that loudspeaker parameters real-time estimation obtains can be hindered into Re, loudspeaker Impedance curve Z and loudspeaker resonant frequency f0Be transferred to Working Status Monitoring subelement 60410, with corresponding threshold value into Row compares, and is monitored in real time with the working condition to loudspeaker, to ensure the normal operation of loudspeaker, the technology of this field Personnel can be configured according to the actual needs, and the present invention is herein with no restrictions.
Those of ordinary skill in the art will appreciate that all or part of the steps in the various methods of above-described embodiment is can It is completed with instructing relevant hardware by program, which can store in computer readable storage medium, and storage is situated between Matter may include: ROM, RAM, disk or CD etc..
The method and system of the embodiment of the present invention are had been described in detail above, the present invention is not limited thereto.Any Field technical staff can make various changes or modifications without departing from the spirit and scope of the present invention, therefore guarantor of the invention Shield range should be defined by the scope defined by the claims..

Claims (16)

1. a kind of speaker control method characterized by comprising
Obtain the digital signal of input;
The digital signal of input is filtered and buffered respectively, obtains the buffering signals at current time;
Based on the voltage signal and current signal at loudspeaker both ends described in current time, the gain of the buffering signals is adjusted Section, within a preset range by the structural behaviour state modulator of the loudspeaker, comprising:
Down-sampled processing is carried out to acquired real-time voltage signal and current signal respectively;
Based on the voltage signal and current signal after down-sampled, calculate separately to obtain what current time controlled voice coil temperature Temperature gain coefficient and the displacement gain coefficient that vibrating diaphragm displacement is controlled;
The maximum permissible voltage of the power amplifier of voltage and the loudspeaker based on the buffering signals, is calculated to institute State the signal gain coefficient that the voltage of buffering signals is controlled;
Based on the signal gain coefficient, temperature gain coefficient and displacement gain coefficient, the buffering signals are calculated most Whole gain coefficient;
Gain process is carried out to the buffering signals using the final gain coefficient.
2. speaker control method according to claim 1, which is characterized in that the structural behaviour parameter packet of the loudspeaker Include at least one of voice coil temperature and vibrating diaphragm displacement.
3. speaker control method according to claim 2, which is characterized in that the voltage signal based on after down-sampled And current signal, the temperature control gain coefficient is calculated, comprising:
According to the voltage signal and current signal after down-sampled, the resistance of voice coil direct current is calculated;
Using the voice coil direct current resistance being calculated, the voice coil temperature at current time is calculated;
The maximum allowable temperature of voice coil temperature and the voice coil based on current time, is calculated the temperature gain coefficient.
4. speaker control method according to claim 2, which is characterized in that the voltage signal based on after down-sampled And current signal, the displacement gain coefficient at current time is calculated, comprising:
Using the voltage signal and current signal after down-sampled, the impedance transfer function at current time is calculated;
According to the impedance transfer function, the electric voltage displacement transmission function of the loudspeaker is determined;
According to the electric voltage displacement transmission function of the loudspeaker and it is down-sampled after buffering signals, the vibration at current time is calculated Film displacement;
The maximum allowable displacement of vibrating diaphragm displacement and the vibrating diaphragm based on current time, is calculated the displacement gain coefficient.
5. speaker control method according to claim 4, which is characterized in that carried out in the input signal to current time When filtering processing, the centre frequency of trapper is resonance frequency of the loudspeaker at current time in used filter group Rate.
6. speaker control method according to claim 5, which is characterized in that the resonance frequency at the loudspeaker current time Rate is calculated using following formula:
Wherein, f0(n) indicate that the resonant frequency at the n-th moment of loudspeaker, Z (n, f) are frequency The impedance transfer function in domain, max () indicate maximum value, and abs () indicates the amplitude of plural number.
7. speaker control method according to claim 6, which is characterized in that it is described it is down-sampled after voltage signal and electricity The sample rate of stream signal meets following formula:
4·f0≤fs_ctrl≤10·f0, wherein fs_ctrlThe sample rate of voltage signal and current signal after indicating down-sampled, f0 Indicate the loudspeaker in the resonant frequency at current time.
8. speaker control method according to claim 2, which is characterized in that calculate the buffering using following formula The final gain coefficient of signal:
gtot(n)=min (gs(n),gx(n))*gT(n), wherein gtot(n) the final increasing of buffering signals described in current time is indicated Beneficial coefficient, gs(n) the signal gain coefficient, g are indicatedT(n) the temperature gain coefficient, g are indicatedx(n) indicate that the displacement increases Beneficial coefficient.
9. a kind of loudspeaker controller characterized by comprising
Acquiring unit, suitable for obtaining the digital signal of input;
Filter processing unit is filtered respectively suitable for the digital signal to input;
Buffered unit obtains buffering signals suitable for carrying out buffered to the signal after filtering processing;
Gain processing unit, suitable for voltage signal and current signal based on loudspeaker both ends described in current time, to described slow The gain for rushing signal is adjusted, within a preset range by the structural behaviour state modulator of the loudspeaker, comprising:
Down-sampled subelement, suitable for carrying out down-sampled processing to acquired real-time voltage signal and current signal respectively;
Temperature controls subelement, suitable for current time is calculated to sound based on the voltage signal and current signal after down-sampled The temperature gain coefficient that circle temperature is controlled;
Bit andits control subelement, suitable for current time is calculated to vibration based on the voltage signal and current signal after down-sampled Film is displaced the displacement gain coefficient controlled;
Signal controls subelement, and the maximum of the power amplifier suitable for voltage and the loudspeaker based on the buffering signals permits Perhaps the signal gain coefficient controlled the voltage of the buffering signals is calculated in voltage;
Computation subunit is suitable for being based on the signal gain coefficient, temperature gain coefficient and displacement gain coefficient, institute is calculated State the final gain coefficient of buffering signals;
Gain process subelement is suitable for carrying out gain process to the buffering signals using the final gain coefficient.
10. loudspeaker controller according to claim 9, which is characterized in that the structural behaviour parameter of the loudspeaker At least one of be displaced including voice coil temperature and vibrating diaphragm.
11. loudspeaker controller according to claim 10, which is characterized in that the temperature control subelement is suitable for root According to the voltage signal and current signal after down-sampled, the resistance of voice coil direct current is calculated;Using the voice coil direct current resistance being calculated, meter Calculate the voice coil temperature at current time;The maximum allowable temperature of voice coil temperature and the voice coil based on current time, is calculated The temperature gain coefficient.
12. loudspeaker controller according to claim 10, which is characterized in that the Bit andits control subelement is suitable for adopting With the voltage signal and current signal after down-sampled, the impedance transfer function at current time is calculated;It is passed according to the impedance Delivery function determines the electric voltage displacement transmission function of the loudspeaker;According to the electric voltage displacement transmission function and drop of the loudspeaker The vibrating diaphragm displacement at current time is calculated in buffering signals after sampling;Vibrating diaphragm displacement and the vibrating diaphragm based on current time Maximum allowable displacement, the displacement gain coefficient is calculated.
13. loudspeaker controller according to claim 12, which is characterized in that the filter processing unit is to current When the input signal at moment is filtered, the centre frequency of trapper is that the loudspeaker is worked as in used filter group The resonant frequency at preceding moment.
14. loudspeaker controller according to claim 13, which is characterized in that the resonance at the loudspeaker current time Frequency is calculated using following formula:
Wherein, f0(n) indicate that the resonant frequency at the n-th moment of loudspeaker, Z (n, f) are frequency The impedance transfer function in domain, max () indicate maximum value, and abs () indicates the amplitude of plural number.
15. loudspeaker controller according to claim 14, which is characterized in that it is described it is down-sampled after voltage signal and The sample rate of current signal meets following formula:
4·f0≤fs_ctrl≤10·f0, wherein fs_ctrlThe sample rate of voltage signal and current signal after indicating down-sampled, f0 Indicate the loudspeaker in the resonant frequency at current time.
16. loudspeaker controller according to claim 10, which is characterized in that the computation subunit is suitable for using such as Under formula calculate the final gain coefficients of the buffering signals:
gtot(n)=min (gs(n),gx(n))*gT(n), wherein gtot(n) the final increasing of buffering signals described in current time is indicated Beneficial coefficient, gs(n) the signal gain coefficient, g are indicatedT(n) the temperature gain coefficient, g are indicatedx(n) indicate that the displacement increases Beneficial coefficient.
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