CN201160320Y - Novel DSP automatic notch digital frequency shifter - Google Patents
Novel DSP automatic notch digital frequency shifter Download PDFInfo
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- CN201160320Y CN201160320Y CNU2007200509229U CN200720050922U CN201160320Y CN 201160320 Y CN201160320 Y CN 201160320Y CN U2007200509229 U CNU2007200509229 U CN U2007200509229U CN 200720050922 U CN200720050922 U CN 200720050922U CN 201160320 Y CN201160320 Y CN 201160320Y
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Abstract
The utility model relates to a novel DSP automatic notch digital frequency shifter, which is characterized in that: the frequency shifter comprises an audio signal input circuit 1, a modulus converter 2, a DSP digital signal processor 3, a band-pass filter circuit 4 and an output amplifier circuit 5; the output end of the audio signal input circuit 1 is connected with the input end of the modulus converter 2, the audio input/output port of the DSP digital signal processor 3 is connected with the input/output port of the audio signal input circuit 1, the audio signal output end of the DSP digital signal processor 3 is outputted by an amplifying circuit 5 and fed back to the feedback input end of the modulus converter 2 by the band-pass filter circuit. The novel DSP automatic notch digital frequency shifter adopts the DSP digital signal processor so as to eliminate and solve the self-exciting howl of microphone resulted from the echo of loudspeaker, thereby improving gain of microphone and meeting the requirement of sound enlarging in various occasions.
Description
Technical field
The utility model relates to the automatic trap numeral of a kind of New DSP frequency shifter, is a kind of professional public address equipment DSP numeral frequency shifter that suppresses acoustic feedback.Belong to the Audio Signal Processing field.
Background technology
It is relatively stubborn problem of sound equipment circle that acoustic feedback in the public address engineering suppresses.Existing frequency shifter is to adopt the simulation frequency shift technique to realize, generally be to form by band pass filter circuit, modulator circuit, demodulator circuit, owing to modulator, demodulator carrier phase difference or difference on the frequency are realized the audio frequency shift frequency, this analoglike frequency shifter is made of discrete component, has the shortcoming of big, the anti-feedback weak effect of the distortion factor.It is that " 02223004.1 ", name are called the utility model patent of " a kind of shift frequency amplifier " that the Chinese patent communique discloses application number, and its design feature is: be made up of pre-amplification circuit, band pass filter circuit, modulator circuit, demodulator circuit, carrier oscillator circuit, output amplifier circuit etc.Though the anti-feedback effect that this shift frequency increases chapter device discrete component frequency shifter than before has improvement, its integrated level to increase, simulate and understand still in the frequency shifter practical application that frequency of occurrences response is narrower, the distortion factor is big, components and parts multicircuit complexity, produce debugging complexity, poor heat stability, be subject to problem such as radio frequency interference.
The utility model content
The purpose of this utility model, be for the automatic trap of a kind of New DSP numeral frequency shifter is provided, it has anti-ly utters long and high-pitched sounds byer force, and Hz-KHz is wideer, the distortion factor is littler, can better eliminate and solve loud speaker and return the microphone self-excitation that is subjected to and causes and utter long and high-pitched sounds, improve the characteristics of microphone gain.
The purpose of this utility model can reach by following measure:
The automatic trap numeral of a kind of New DSP frequency shifter is characterized in that: comprise audio signal input circuit, analog to digital converter, DSP digital signal processor, band pass filter circuit and output amplifier circuit; The input of the output connection mode number converter of audio signal input circuit, the audio frequency I/O port of DSP digital signal processor is connected with the I/O port of audio signal input circuit, and the audio signal output end of DSP digital signal processor reaches the feedback input end that feeds back to analog to digital conversion circuit by bandwidth-limited circuit by output amplifier output.
The audio signal input circuit to the simulated audio signal of system input decay, amplitude limit, make it to mate with the analog to digital converter incoming signal level; Analog to digital converter carries out analog-to-digital conversion to simulated audio signal and exports with the digital audio-frequency data form; DSP digital signal processor word audio signal is analyzed, shift frequency and trap are handled, and converts simulated audio signal output to; Band pass filter circuit receive from the simulated audio signal of DSP digital signal processor output gain and bandpass filtering treatment after feed back to the feedback input end of analog to digital conversion circuit; Output amplifier circuit receives from the simulated audio signal of DSP digital signal processor output, gain and bandpass filtering treatment after deliver to audio output.
The purpose of this utility model can also reach by following measure:
A kind of execution mode of the present utility model is: described DSP digital signal processor is made up of two-way digital to analog converter, voice data interface, arithmetic element, microcontroller interface, dynamic memory, parameter storage, command processor and digital filter; The voice data interface is connected with analog to digital converter, the two-way digital to analog converter is connected with output amplifier with band pass filter circuit respectively, and the input of microcontroller interface is connected with the I/O mouth of microprocessor, its output is connected with arithmetic element by command processor.
A kind of execution mode of the present utility model is: also comprise microprocessor, display unit circuit and button inputting circuits, the I/O mouth of described microprocessor is connected with the I/O mouth of DSP digital signal processor, the signal output part of microprocessor connects the input of display unit circuit, and the signal output part of microprocessor connects the output of button inputting circuits.
A kind of execution mode of the present utility model is: also comprise the signal processing circuit of uttering long and high-pitched sounds, the input of the described signal processing circuit of uttering long and high-pitched sounds is connected with the output of output amplifier circuit, and the output of the signal processing circuit of uttering long and high-pitched sounds connects the input of microprocessor.
A kind of execution mode of the present utility model is: described display unit circuit mainly is formed by connecting by triode Q 1～Q4, LED1～LED8, charactron DIGI, shift register chip IC 16 and peripheral resistance and electric capacity thereof; Button inputting circuits is formed by connecting by button K1, K2, K3 and interface CN1.
A kind of execution mode of the present utility model is: described audio frequency input circuit is made up of resistance R 21～R23, capacitor C 20～C21, diode D10～D11; Form attenuator circuit by R21, R22, form amplitude limiter circuit by D10, D11.
A kind of execution mode of the present utility model is: described band pass filter circuit is made up of resistance R 11～R14, capacitor C 10～C14 and integrated operational amplifier chip U2.
A kind of execution mode of the present utility model is: described output amplifier is made up of R71～R74, capacitor C 70～C52 and integrated operational amplifier chip U3.
A kind of execution mode of the present utility model is: the described signal processing circuit of uttering long and high-pitched sounds is made up of R1～R6, capacitor C 1～C3, diode D9, triode Q9 and integrated operational amplifier chip U9.
Principle of the present utility model: adopt A/D converter that the input simulated audio signal is carried out A/D and convert digital signal to, be sent to DSP digital signal processor inside audio digital signals is analyzed and shift frequency and trap processing, convert simulated audio signal output to by D/A converter again.The DSP digital signal processor is connected with microprocessor, and microprocessor control DSP handles the digital audio and video signals parameters, and microprocessor search Audio squealing point, and send DSP the point data of uttering long and high-pitched sounds, and DSP handles digital audio and video signals and finishes a trap of uttering long and high-pitched sounds.
The course of work of the present utility model:
The external analog audio signal is imported from audio input interface, enters the audio frequency input circuit the signal amplitude limiting processing that decays.
Simulated audio signal after the processing enters from the analog to digital converter input port, and simulated audio signal carries out exporting with the digital audio-frequency data form after the analog-to-digital conversion.
The analog-digital conversion data mouth connects DSP internal data interface, DSP internal arithmetic unit carries out shift frequency computing, high pass adjustment computing, low pass adjustment computing, tone processing computing, entire gain calculation process etc. to digital audio and video signals, digital to analog converter by DSP inside is converted to simulated audio signal, exports from delivery outlet.
Simulated audio signal enter again band pass filter be with logical and gain process after, 1 input from the analog to digital converter input port, analog to digital converter carries out once more analog-to-digital conversion to analogue audio frequency and becomes the output of digital audio-frequency data form, be sent to the DSP inter-process by the DSP data-interface, the DSP digital signal processor carries out digital notch according to the data cases of the dot frequency of uttering long and high-pitched sounds to digital audio-frequency data to be handled.
The inner digital to analog converter of DSP is converted to simulated audio signal with digital audio and video signals, and from 2 outputs of digital-to-analogue conversion delivery outlet, the simulated audio signal of output enters output amplifier circuit, and simulated audio signal is through amplifying and being with the logical back of handling to export from audio output port.
Output amplifier circuit output connects the signal processing circuit of uttering long and high-pitched sounds in addition, and the near sinusoidal ripple that will produce when the signal processing circuit of uttering long and high-pitched sounds will be uttered long and high-pitched sounds is converted into square wave output, and connects fracture in the microprocessor; At this moment, the square wave trailing edge produces and interrupts, microprocessor is counted and is analyzed frequency, judge whether to produce and utter long and high-pitched sounds, audio producing positive feedback self-excitation when producing,, microphone position certain and direction one at stereo set and environment because utter long and high-pitched sounds regularly, this frequency is the frequency of a basic fixed in the certain hour, according to these characteristics, microprocessor can judge whether to produce and utter long and high-pitched sounds; When differentiating when uttering long and high-pitched sounds, microprocessor transmits these frequency trap data to DSP, and handle again to the frequency trap data DSP inside, and trap can be handled the gain of the dot frequency of uttering long and high-pitched sounds automatically, and the ensemble average that its result has just improved audio frequency greatly gains.
The beneficial effects of the utility model are:
1, the utility model adopts the DSP digital signal processor, analog to digital converter adopts the 24bit96kHzA/D conversion, digital to analog converter adopts 1-bit ∑ Δ type D/A conversion, make that the audio frequency frequency response is wide, can reach 60Hz～20kHz, shift frequency amount adjustable range-16Hz～+ 16Hz, high pass adjustable range 60Hz～300Hz, low pass adjustable range 8kHz～20kHz, but and more than 3 of memory storage utter long and high-pitched sounds a little, and the dot frequency signal of uttering long and high-pitched sounds reduced 4dB.
2, the utility model has increased input of microprocessor, keyboard and display unit circuit, uses software to control the frequency shifter parameters, and search is uttered long and high-pitched sounds a little and to a trap of uttering long and high-pitched sounds automatically; Can the flexible configuration display unit, parameter software control, and parameter stored automatically can better be eliminated and solve loud speaker and be returned the microphone self-excitation that is subjected to and causes and utter long and high-pitched sounds, and improves microphone gain, satisfies the requirement of various occasion sound equipment public address.
Description of drawings
Fig. 1 is a circuit structure block diagram of the present utility model.
Fig. 2 is an audio frequency input circuit schematic diagram of the present utility model.
Fig. 3 is a band pass filter circuit schematic diagram of the present utility model.
Fig. 4 is an output amplifier schematic diagram of the present utility model.
Fig. 5 is the signal processing circuit schematic diagram of uttering long and high-pitched sounds of the present utility model.
Fig. 6 is display circuit of the present utility model and button input figure.
Fig. 7 is a microprocessor program flow chart of the present utility model.
Embodiment
Specific embodiment 1:
With reference to Fig. 1, present embodiment comprises audio signal input circuit 1, analog to digital converter 2, DSP digital signal processor 3, band pass filter circuit 4 and output amplifier circuit 5; The input of the output connection mode number converter 2 of audio signal input circuit 1, the audio frequency I/O port of DSP digital signal processor 3 is connected with the I/O port of audio signal input circuit 1, and DSP digital signal processor 3 audio signal output ends reach the feedback input end that feeds back to analog to digital conversion circuit 2 by bandwidth-limited circuit 4 by output amplifier 5 outputs.
With reference to Fig. 2, audio frequency input circuit 1 is by resistance R 21, R22, R23; Capacitor C 20, C21; Diode D10, D11 form.The effect of this circuit be to the simulated audio signal of system input decay, amplitude limit, make simulated audio signal and analog to digital converter incoming signal level coupling.R21, R22 form attenuator circuit, and D10, D11 form amplitude limiter circuit.
In the present embodiment: analog to digital converter 2 adopts 24bit96kHz A/D conversion.DSP digital signal processor 3 adopts the TC9444F of TOSHIBA company, its inside comprises a two-way digital to analog converter 31, voice data interface 32, addition multiplying unit 33, microprocessor and connects parts such as 34,64Kbit dynamic memories 35, parameter storage 36, command processor 37, a digital filter 38, is used for the processing of digital voice data.Its inner two-way D/A converter adopts 1-bit ∑ Δ number of types weighted-voltage D/A converter, acoustical sound when guaranteeing digital-to-analogue conversion.Its voice data interface can send, receive the IIS voice data, also can be with MASTER/SLAVE mode transfer voice data.
" E " some input of simulated audio signal from Fig. 2 from audio frequency input IN input, through overdamping, export from " F " point after the amplitude limiting processing, signal after the processing enters from the input port 2 of analog to digital converter 2, carry out analog-to-digital conversion, output to DSP digital signal processor 3 with the digital audio-frequency data form then, DSP internal arithmetic unit carries out the shift frequency computing to digital audio and video signals, high pass is adjusted computing, low pass is adjusted computing, tone is handled computing, after the entire gain calculation process, digital to analog converter by DSP inside is converted to simulated audio signal, exports from delivery outlet OT after output amplifier amplifies.
With reference to Fig. 3, band pass filter circuit 4 is by resistance R 11, R12, R13, R14; Capacitor C 10, C12, C13, C14; Integrated operational amplifier U2 forms (as Fig. 3).The effect of this circuit is the simulated audio signal that receives from digital to analog converter delivery outlet-output, gains and bandpass filtering treatment, sends into the analog to digital converter input port after the output again.
Deliver to C point among Fig. 3 by the signal of the delivery outlet of DSP digital signal processor 3 output, through the output of the D point from Fig. 3 after gain and the bandpass filtering treatment, input port from analog to digital converter 2 enters again, carry out analog-to-digital conversion once more, and output to DSP digital signal processor 3 with the digital audio-frequency data form, digital audio and video signals carries out after digital notch handles, and is converted to simulated audio signal by the digital to analog converter of DSP inside, exports from delivery outlet again.
With reference to Fig. 4, output amplifier 5 is by R71, R72, R73, R74; Capacitor C 70, C71, C72, C73; Integrated operational amplifier U3 forms.The effect of this circuit is the simulated audio signal that receives from two outputs of digital to analog converter delivery outlet, gains and bandpass filtering treatment, outputs to utter long and high-pitched sounds signal processing circuit and system audio delivery outlet respectively.
With reference to Fig. 5, the signal processing circuit of uttering long and high-pitched sounds 9 is by R1, R2, R3, R4, R5, R6; Capacitor C 1, C2, C3; Diode D1; Triode Q1, integrated operational amplifier U 1 forms.Public address is certain frequency and fluctuation is very little, big near sinusoidal ripple during the ratio of gains operate as normal when uttering long and high-pitched sounds and producing.When the effect of this circuit produces when uttering long and high-pitched sounds exactly, receive from the next sinusoidal wave audio frequency signal of output amplifier, be converted into square wave, send fracture processing in the microprocessor from " Z " point.
Deliver to Y point Fig. 4 from the signal of DSP digital signal processor 3 delivery outlets output, after gain and bandpass filtering treatment, the Z point of signal from Fig. 4 outputs to the A point among Fig. 5, and the near sinusoidal ripple that will produce when the signal processing circuit of uttering long and high-pitched sounds 9 will be uttered long and high-pitched sounds is converted into behind the square wave B point from Fig. 5 and exports.
With reference to Fig. 6, display unit circuit 7 and button inputting circuits 8 are by resistance R 1, R2, R3, R4, R5, R6, R7, R8, R9, R10, R11, R12, R13, R14, R15, R16, R17, R18; Capacitor C 1; LED 1, LED2, LED3, LED4, LED5, LED6, LED7, LED8; Charactron DIG1; Triode Q1, Q2, Q3, Q4; Button K1, K2, K3; Shift register integrated circuit (IC) 16; Interface CN1 forms.1～9 pin of CN1 connects the I/O mouth of microprocessor.Button K1, K2, K3 are connected the I/O mouth of microprocessor, are high level states when not having button to press, and become low level when button is pressed, and the microprocessor internal program has judged whether that thus key presses.Shift register integrated circuit (IC) 16 is that a serial data is changeed the parallel data integrated circuit, its data port is connected the I/O mouth of microprocessor, what microprocessor was taked is the working method of dynamic refresh, content displayed is sent to shift register integrated circuit (IC) 16 with serial data mode as required, IC16 exports with parallel mode, send several the time to IC16, microprocessor is controlled the base stage level of Q1, Q2, Q3, Q4 successively and is implemented in conducting in turn, the section or the light-emitting diodes pipeline section of the charactron that gating will show.Realize the demonstration of dynamic refresh mode.
Microprocessor 6 adopts the ST89C52RC of STC Corporation, is the double-speed 8051 kernel microprocessors of 6 a clocks/machine cycle, but characteristics such as high speed, low-power consumption, wide voltage ISP/IAP programming are arranged.Its built-in 8K byte FLASH memory is used for stored program and fixed data, and built-in 2K byte can be used for storage and memory audio frequency parameter etc. by electricity rewriting eeprom memory, and has simplified the external circuit structure.The effect one of microprocessor is by transmitting control command to DSP, can controlling DSP internal work pattern and voice data processing method, as shift frequency number, high pass value, low-pass value, the trap value etc. of uttering long and high-pitched sounds.Whether effect two is can connect button at the I/O mouth to form keyboard circuit, differentiate button and press, and internal processes carries out respective handling.Effect three is that the I/O mouth connects display circuit, shows the parameter values of the current set of menu or adjustment, handled easily person's operation.
The square wave of the B point output from Fig. 5 enters fracture in the microprocessor, and microprocessor is counted and analyzed frequency, judges whether to produce and utters long and high-pitched sounds, and does not utter long and high-pitched sounds if do not produce, and directly the output of the Z point from Fig. 4 is through the voice signal of shift frequency; Utter long and high-pitched sounds if produce, microprocessor transmits these frequency trap data to DSP, handle again to the frequency trap data DSP inside, trap can be handled the gain of the dot frequency of uttering long and high-pitched sounds automatically, digital to analog converter by DSP inside is converted to simulated audio signal again, output to output amplifier 5 from delivery outlet 2, through gain and bandpass filtering treatment, the Z point output among last Fig. 4 is through the voice signal of shift frequency.
With reference to Fig. 7, the microprocessor program flow process is:
Microprocessor selects for use inside to have the microprocessor of EEPROM, can rewrite and store the mnemon content.
Program is at first carried out initialization to the working method and the external pin of internal register, open external interrupt and total the interruption, control reset DSP and AD converter, read inner EEPROM then about entire gain value, high pass parameter value, low pass parameter value, shift frequency numerical value, high pitch pitch value, bass pitch value, a plurality of dot frequency value of uttering long and high-pitched sounds, then the parameter control data is transmitted DSP inside, DSP internal command processor is adjusted inner workings and data operation mode.
The microprocessor main flow begins to have judged whether that button presses, if there is button to press, differentiation is what button and handling respectively, if the function selecting button is pressed, program switches to the function of next functional parameter.Present embodiment tunable integer body yield value, high pass parameter value, low pass parameter value, shift frequency numerical value, high pitch pitch value, bass pitch value, a plurality of dot frequency value etc. of uttering long and high-pitched sounds.Can adjust numerical value by the plus-minus button when entering each function.Do not press if there is button, carry out the further work-discriminating program of uttering long and high-pitched sounds.
The discriminating program of uttering long and high-pitched sounds judges that current whether the generation utter long and high-pitched sounds, when detecting approximate single-frequency, a period of time, promptly remembers current frequency numerical value this moment for uttering long and high-pitched sounds, and trap control data transmission DSP, as the nothing generation of uttering long and high-pitched sounds, carry out further work-demonstration refurbishing procedure.
Show the difference of refurbishing procedure, and take different control and data mode according to outernal display unit.Because the present embodiment adjustable parameters is a lot, numerical value that display routine meeting dynamic refresh shows and unit etc. are operated with the handled easily person.
Continue to return button and judge that main flow is finished.
DSP of the present utility model numeral frequency shifter is applied to the inhibitor or DSP numeral frequency shifter is installed of uttering long and high-pitched sounds in KTV, meeting, performance, the academic report Room in the public address power amplifier, reach very good acoustic feedback and suppressed effect, can improve mike gain 6～12dB, acoustical quality is good, frequency response reaches 60Hz～20kHz, imperceptible audio distortion, key controlled and demonstration feels easy to use.
Claims (9)
1, the automatic trap numeral of New DSP frequency shifter is characterized in that: comprise audio signal input circuit (1), analog to digital converter (2), DSP digital signal processor (3), band pass filter circuit (4) and output amplifier circuit (5); The input of the output connection mode number converter (2) of audio signal input circuit (1), the audio frequency I/O port of DSP digital signal processor (3) is connected with the I/O port of audio signal input circuit (1), and DSP digital signal processor (3) audio signal output end reaches the feedback input end that feeds back to analog to digital conversion circuit (2) by bandwidth-limited circuit (4) by output amplifier (5) output.
2, the automatic trap numeral of New DSP according to claim 1 frequency shifter, it is characterized in that: described DSP digital signal processor (3) is made up of two-way digital to analog converter (31), voice data interface (32), arithmetic element (33), microcontroller interface (34), dynamic memory (35), parameter storage (36), command processor (37) and digital filter (38); Voice data interface (32) is connected with analog to digital converter (2), the two-way digital to analog converter is connected with output amplifier (5) with band pass filter circuit (4) respectively, and the input of microcontroller interface (34) is connected with the I/O mouth of microprocessor (6), its output is connected with arithmetic element (33) by command processor (37).
3, the automatic trap numeral of New DSP according to claim 1 frequency shifter, it is characterized in that: also comprise microprocessor (6), display unit circuit (7) and button inputting circuits (8), the I/O mouth of described microprocessor (6) is connected with the I/O mouth of DSP digital signal processor (3), the signal output part of microprocessor (6) connects the input of display unit circuit (7), and the signal output part of microprocessor (6) connects the output of button inputting circuits (8).
4, the automatic trap numeral of New DSP according to claim 1 frequency shifter, it is characterized in that: also comprise the signal processing circuit of uttering long and high-pitched sounds (9), the input of the described signal processing circuit of uttering long and high-pitched sounds (9) is connected with the output of output amplifier circuit (5), and the output of the signal processing circuit of uttering long and high-pitched sounds (9) connects the input of microprocessor (6).
5, the automatic trap numeral of New DSP according to claim 3 frequency shifter, it is characterized in that: described display unit circuit (7) mainly is formed by connecting by triode Q1～Q4, LED1～LED8, charactron DIGI, shift register chip IC 16 and peripheral resistance and electric capacity thereof; Described button inputting circuits (8) is formed by connecting by button K1, K2, K3 and interface CN1.
6, the automatic trap numeral of New DSP according to claim 1 frequency shifter, it is characterized in that: described audio frequency input circuit (1) is made up of resistance R 21～R23, capacitor C 20～C21, diode D10～D11; Form attenuator circuit by R21, R22, form amplitude limiter circuit by D10, D11.
7, the automatic trap numeral of New DSP according to claim 1 frequency shifter, it is characterized in that: described band pass filter circuit (4) is made up of resistance R 11～R14, capacitor C 10～C14 and integrated operational amplifier chip U2.
8, the automatic trap numeral of New DSP according to claim 1 frequency shifter, it is characterized in that: described output amplifier (5) is made up of R71～R74, capacitor C 70～C52 and integrated operational amplifier chip U3.
9, the automatic trap numeral of New DSP according to claim 4 frequency shifter, it is characterized in that: the described signal processing circuit of uttering long and high-pitched sounds (9) is made up of R1～R6, capacitor C 1～C3, diode D9, triode Q9 and integrated operational amplifier chip U9.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNU2007200509229U CN201160320Y (en) | 2007-04-28 | 2007-04-28 | Novel DSP automatic notch digital frequency shifter |
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| Application Number | Priority Date | Filing Date | Title |
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| CNU2007200509229U CN201160320Y (en) | 2007-04-28 | 2007-04-28 | Novel DSP automatic notch digital frequency shifter |
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Cited By (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102035775A (en) * | 2009-09-29 | 2011-04-27 | 创杰科技股份有限公司 | Frequency shifter |
| CN102316395A (en) * | 2010-07-09 | 2012-01-11 | 廖明忠 | Method and device for judging and eliminating howlround |
| CN110191398A (en) * | 2019-05-17 | 2019-08-30 | 深圳市湾区通信技术有限公司 | Suppressing method, device and the computer readable storage medium uttered long and high-pitched sounds |
| CN111182431A (en) * | 2019-12-27 | 2020-05-19 | 中山大学花都产业科技研究院 | Howling suppression method for conference sound reinforcement system |
-
2007
- 2007-04-28 CN CNU2007200509229U patent/CN201160320Y/en not_active Expired - Fee Related
Cited By (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102035775A (en) * | 2009-09-29 | 2011-04-27 | 创杰科技股份有限公司 | Frequency shifter |
| CN102035775B (en) * | 2009-09-29 | 2014-02-12 | 创杰科技股份有限公司 | Frequency shifter |
| CN102316395A (en) * | 2010-07-09 | 2012-01-11 | 廖明忠 | Method and device for judging and eliminating howlround |
| CN102316395B (en) * | 2010-07-09 | 2014-12-31 | 深圳市宇恒互动科技开发有限公司 | Method and device for judging and eliminating howlround |
| CN110191398A (en) * | 2019-05-17 | 2019-08-30 | 深圳市湾区通信技术有限公司 | Suppressing method, device and the computer readable storage medium uttered long and high-pitched sounds |
| CN111182431A (en) * | 2019-12-27 | 2020-05-19 | 中山大学花都产业科技研究院 | Howling suppression method for conference sound reinforcement system |
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| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant | ||
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| CF01 | Termination of patent right due to non-payment of annual fee |
Granted publication date: 20081203 Termination date: 20110428 |