111 results about "Sound reinforcement system" patented technology

A sound reinforcement system which enables handsfree and high-quality sound reinforcement without requiring a person who is speaking to move to a microphone or move a microphone. At least one microphone and a plurality of speakers are arranged in a room. A speaker output adjusting section outputs sound picked up by the microphone to the plurality of speakers at predetermined levels.

A computing system automates the configuration and management of a live sound system that includes a processor and memory for building in a GUI of a display a representation of the live sound system for a venue. The system loads a venue template that includes loudspeaker arrays and related properties including a setup configuration of the loudspeaker arrays and tuning data for constituent loudspeakers that are operable to provide an audio coverage pattern for the venue. The system overlays on top of the representation of the loudspeaker arrays a wiring circuit representation indicating interconnections of the loudspeakers that define bandpass inputs for each array. The system generates a plurality of amplifiers in the representation to drive the arrays, and associates amplifier channels of the amplifiers with the bandpass inputs. The amplifier channels include representations of output channels of DSPs coupled with respective amplifier channels. The system loads tuning data into respective representation of the DSP and / or amplifiers based on configurations of the associated loudspeakers to complete virtual configuration. The representations of the devices and connections may be matched with physical devices of the live sound system and the tuning data sent down to the physical DSPs and / or amplifiers for their configuration.

A sound reinforcement system which enables handsfree and high-quality sound reinforcement without requiring a person who is speaking to move to a microphone or move a microphone. At least one microphone and a plurality of speakers are arranged in a room. A speaker output adjusting section outputs sound picked up by the microphone to the plurality of speakers at predetermined levels.

The invention relates to a method and a device for detecting the howling frequency point of an acoustic amplification system. The method comprises the following steps of converting collected voice signals to frequency domain signals; searching for sub-bands where the howling frequency point exists within the bandwidth range of the frequency domain signals; detecting the howling frequency point within the range of the sub-bands where the howling frequency point exists. According to the method and the device for detecting the howling frequency point of the acoustic amplification system provided by the invention, the frequency point at which the howling occurs can be precisely and automatically positioned by compressing the searching range of the howling frequency point twice and by sampling and extracting time domain signals according to the howling frequency point existing range; the method is low in complexity and convenient to realize in real time on hardware, so that the howling inhibition of the acoustic amplification system can be further realized more economically.

The invention discloses a self-adaptive adjustment method for an indoor sound reinforcement. The self-adaptive adjustment method comprises a first step of adjusting an input gain; a second step of testing an upper limit of an output gain; a third step of adjusting the output gain; a fourth step of detecting output delay and conducting delay compensation; a fifth step of detecting a frequency response and conducting frequency compensation; a sixth step of detecting various parameters of the system after adjustment; and a seventh step of outputting adjustment results. The scheme achieves automatic adjustment for the sound reinforcement system with no human intervention and good consistency after adjustment. The invention is applicable to all indoor sound reinforcement systems.

The invention discloses a method for detecting an indoor sound reinforcement device. The method comprises a first step that a DSP processing module controls a controllable output matrix to output a test signal to a speaker array; a second step that a sound pick-up array collects an indoor sound signal and sends the indoor sound signal to an audio analysis module; a third step that the audio analysis module analyzes loudness, phase, frequency response and distortion factor of the sound signal; a fourth step that the DSP processing module calculates a balance degree of the sound field in an indoor environment according to the loudness, calculates delay according to the phase difference, and calculates the matching of the sound built environment and the sound reinforcement device according to the frequency response and the distortion factor; and a fifth step that test results are outputted. The automatic detection function for the indoor sound reinforcement device can be used for bringing standardized and quantified acceptance criteria to acceptance of project and greatly simplifying the acceptance process. The invention is applicable to all indoor sound reinforcement systems.

The invention relates to an onsite classroom pickup and loudspeaking system adapted to the position of a speaker. The system comprises a pickup unit, a loudspeaker array and a processing circuit, wherein the pickup unit converts obtained audio signals of the speaker into electrical signal based on the position of the speaker, the processing circuit removes noises of the electrical signals and amplifies the de-noised signals to generate driving signals, and the loudspeaker array plays sounds of the speaker according to the driving signals. The system of the invention needs only one pickup unit, the speaker can walk in the classroom to teach class needless of wearing extra devices, local environmental noises can be effectively inhibited, and the problem of howling is eliminated.

The present invention discloses an adaptive sub-band audio feedback suppression method. The method comprises the steps of segmenting an input signal according to a sampling sequence to obtain data blocks; allowing the data blocks to pass a sub-band filter to obtain sub-band signals of a power amplification signal and an input signal; allowing the sub-band signals to pass an adaptive sub-band feedback filter to deduce a microphone feedback sub-band signal generated by a speaker signal, performing weighing energy analysis and statistics on the sub-band signal, and correcting the adaptive sub-band feedback filter; synthesizing input sub-band signals expect the feedback sub-band signal and recovering to obtain a time domain signal; and sending the recovered time domain signal to the speaker for sound amplification. The method has the advantages that the signal filter is segmented into sub-bands, so that the correlation between the sub-band signals is lowered, a convergence problem of the adaptive filter is solved, data processing amount of the system is reduced, the feedback suppression processing efficiency is improved, the sound gain of the sound amplification system is improved, the sound amplification quality is improved, and the sound gain can be increased by over 6 decibel stably, at most 15 decibel.

The invention relates to an audio dynamic feedback suppression method, which relates to the field of digital audio processing in a sound reinforcement system. The prior art has the defects of slow response speed, low resolution and the like. The method comprises the following steps: decomposing digital audio signals into sub signals at a plurality of frequency bands; according to the preset sampling rate, desampling the signal at each frequency band after being filtered; carrying out the fast Fourier transform analysis for digital audio at each frequency band after being desampled, and estimating the self-excitation frequency; and filtering the self-excitation frequency according to the estimated result. Under the premise of ensuring good robustness, the response speed and resolution are improved, and the filtering is precise.

The present invention discloses a method and device for controlling speaker array sound field based on a quadratic residue sequence combination. The method comprises steps of: (1) fragmenting a designated quadratic residue sequence in terms of the number of array elements, to generate a plurality of quadratic residue subsequences; (2) designing an optimal array phase delay vector utilizing these subsequences; (3) controlling transmission signals of multi-element channels according to the optimal phase delay vector to adjust phase delay; (4) sending the multi-channel signals subjected to adjustment to a multi-channel power amplifier, to drive the speaker array to generate uniform sound field. The device comprises a sound source, an optimal phase delay estimator, an optimal phase delay controller, a multi-channel power amplifier and a speaker array. The invention can expand the coverage range of sound field radiated from an array and improve uniformity of the sound field. Furthermore, according to the invention, the hardware implementation of the control method of sound field is simple, and the spatial distribution characteristics of sound field meet the requirements of array sound reinforcement system.

The invention discloses a method and a device for controlling a broadband sound field of a loudspeaker array by utilizing a secondary residual sequence. The method comprises the following steps: carrying out multichannel FFT (fast Fourier transform); carrying out multichannel multi-subband equilibrium processing; carrying out multichannel multi-subband phase delay processing; carrying out IFFT (inverse fast Fourier transform); and sending a multichannel time domain sequence to a multichannel power amplifier to drive the loudspeaker array to generate a uniform sound field. The device comprises a sound source, a multichannel FFT transformer, a multichannel multi-subband equalizer, a parameter estimator for multichannel equilibrium, a multichannel multi-sub-band phase delayer, an optimal phase delay estimator, a multichannel IFFT transformer, the multichannel power amplifier and the loudspeaker array which are successively and sequentially connected. The method and the device can be used for effectively expanding the coverage area of a broadband array space radiation sound field and improving the uniformity degree of the broadband array space radiation sound field, so that the requirement on the sound field space distribution quality of a broadband array sound amplification system is met.

The invention discloses an adaptive gain control method of a sound reinforcement system. The method comprises the following steps: S1, presetting an initial system gain, and directly picking up a played sound reinforcement source signal in a circuit of the sound reinforcement system; S2, multiplying the sound reinforcement source signal by the initial system gain, and then playing the sound reinforcement source signal through a loudspeaker in the sound reinforcement system; S3, picking up an onsite sound reinforcement mixed signal by setting a microphone on a sound reinforcement site, that is,a signal obtained by performing convolution on a product of the sound reinforcement source signal and the system gain with a sound field pulse, and then adding a noise signal; S4, calculating an estimated onsite signal to noise ratio according to a selected function relationship by using an MSC signal to noise characterization value; and S5, adjusting the gain of the sound reinforcement system according to the estimated onsite signal to noise ratio. By adoption of the adaptive gain control method of the sound reinforcement system provided by the invention, the gain setting of the sound reinforcement system is more accurate, the sound reinforcement volume is suitable, and the adjusted sound signal on the sound reinforcement site is obviously clearer than the onsite sound signal performed with no adaptive adjustment without increasing sound pollution.

Methods and systems for assisting tonally-challenged singers. A microphone can be integrated with a sound reinforcement system used in a live performance. The microphone, which can transduce the performers voice, can serve multiple purposes such as, for example, to feed input to the natural ear and to the sound reinforcement system. The processed sound of the performers voice (with fundamental frequencies emphasized) can be mixed into the signal fed to a stage “monitor” speaker facing the performer or a headset worn by the performer.

The invention provides a loudspeaker array sound reinforcement system for inhibiting howling. The system comprises an input signal processing module (1), a filter module (2), a multichannel power amplifier (3) and a loudspeaker array (4); the filter module (2) comprises a parameter setting unit (201), a filter group generation unit (202) and a filter group processing unit (203); the filter group generation unit (202) is used for calculating convolution matrixes Hm and Hn formed by time domain impulse response at each control point of the loudspeaker array (4) according to parameters set by the parameter setting unit (201); calculating matrixes Rb and Rd accordingly; calculating a matrix Q according to an average amplitude response fluctuation parameter expression of an equilibrium control point in a sound reinforcement target area; solving a vector w according to Rb, Rd and Q to ensure a maximum value of time domain acoustic energy contrast control of single point equilibrium; converting the vector w into a filter form; and acquiring filter parameters of the filter group processing unit (203).

3D8 is a stereo acoustic amplifying system. Addition to three main channels, namely left, middle and right channels, of the basic acoustic amplifying system, five effect channels are added, including left and right ambient voice channels, top ambient voice channel and front and back stage ambient voice channels. Three main acoustic amplifying channels, including the sub-low-frequency channel, perform basic speech and music amplification, and five ambient voice channels are used to provide various effect voices of performances such as traditional opera, music opera, modern drama, sketch and like.

The invention provides a conference sound amplification system howling suppression method, and the method adopts a frequency shift method and a multi-all-pass filtering random combination method to destroy a phase condition generated by howling, and adopts three steps to detect a howling signal frequency value in a howling signal filtering method, including FFT preliminary detection, double-amplitude refinement and absolute amplitude calculation comparison.

The invention relates to a multifunctional display system for a pedestrian street, which belongs to the fields of communication, display and media technology and comprises a 3G or 4G superhigh brightness outdoor liquid crystal display, a multifunctional charity benefit box and a supporting frame. The 3G or 4G super brightness outdoor liquid crystal display comprises a liquid crystal panel, a high brightness backlight, an automatic brightness adjustment system, a sound reinforcement system, an automatic temperature adjustment system, a 3G / 4G multifunctional drive plate, a special power supply which can supply 9.6V, 12V, 5V multi-group working voltage and a casing; an LVDS interface on the 3G / 4G multifunctional drive plate is connected with the liquid crystal panel, and a RS232 interface of the 3G / 4G multifunctional drive plate is connected with the automatic brightness adjustment system which controls the high-brightness blacklight through a data line; an audio interface on the 3g / 4G multifunctional drive plate is connected with the sound reinforcement system; the automatic temperature adjustment transmits temperature data to the 3G / 4G multifunctional drive plate for controlling the working of a draught fan; and the special power supply supplies a plurality of groups of working voltage and respectively supplies power for each part. The multifunctional display system for a pedestrian street has the functions of automatic receiving, storage and playing of video and texts, issuing of public information and the like.

The invention discloses a howling suppression method based on autonomous learning and a sound amplification system. The howling suppression method and the sound amplification system solve the problem that manual debugging is needed in a wave trapping method. The method comprises a howling suppression trap updating step, and the howling suppression trap parameter updating step comprises the steps that if the howling frequency is detected, parameters of all second-order IIR traps are set in sequence, and meanwhile the reliability of the howling frequency is updated; if the last wave trap parameter is updated and the howling frequency is detected again, the second-order IIR wave trap parameters are covered in sequence until each second-order IIR wave trap parameter is covered by N0 times, and N0 is greater than or equal to 1; sequencing is carried out according to the peak values of the credibility statistical histogram, the previous maximum N1 new peak value ranges are extracted, the center frequency, the quality factor and the gain are calculated according to credibility weights, and the parameters of the N1 new fixed second-order IIR wave traps are determined; and the parameters of the fixed second-order IIR trap are kept unchanged, and the parameters of the dynamic second-order IIR trap are adjusted in real time according to the howling condition.

A system, apparatus and method for configuring a wireless sound reinforcement system detects a controller and two or more wireless adapters using a router. The controller has a plurality of input channels and output channels. Each wireless adapter is associated with an entertainment device. Channel information about each input channel and output channel is obtained from the controller. Device information is obtained from each entertainment device via the wireless adapter. A network address is assigned to each input channel, output channel and wireless adapter. Each entertainment device is mapped to at least one of the input channels and / or output channels. The device information and channel mappings for the entertainment devices are provided to the controller. Data packets are received from the wireless adapters and the controller. A destination network address of each received data packet is identified and each received data packet is sent to the identified destination network address.

The invention relates to an audio signal processing technology, discloses a sound amplification system howling point detection method based on a neural network, and solves the problems of misjudgmentand missed judgment of a howling point and incapability of accurately tracking and detecting the howling point in the traditional technology. The method comprises the following steps: a, acquiring audio data, performing FFT conversion, and marking howling frequency points to form sample data; b, training a defined neural network model based on the sample data to obtain a howling point recognitionmodel; c, in practical application, taking the frequency domain data of the audio data to be processed after FFT conversion as input, and outputting a howling point frequency identification result through a howling point identification model.

The invention discloses a sound amplification system with a reverberation time measurement function. The system comprises a sound amplification box, a microphone and an audio host. The audio host measures reverberation time by adopting an impulse response integral method in which an exponential swept-frequency signal is used as an excitation sound source; the microphone is used for transmitting anacquired audio signal to the audio host; the audio host performs linear convolution on the acquired audio signal and a reverse swept-frequency signal to obtain an impulse response signal; pulse response signals are enabled to pass through octave filters with different center frequencies and corresponding pulse response signals under different center frequencies are obtained; and finally, reverseintegration is carried out on all the pulse response signals to obtain an energy attenuation curve, sample data in the energy attenuation curve is selected to carry out linear fitting, and reverberation time is calculated according to the slope of a linear fitting straight line. The system has the advantages of strong practicability, high precision, small error, short measurement time, simultaneous measurement of a plurality of different positions, remote measurement and the like.

The invention discloses a liquid crystal television system capable of being used under an outdoor environment, belonging to the technical field of radiated televisions and media and comprising a liquid crystal television panel, a liquid crystal television driving plate, an LED backlight system, an automatic brightness regulating system, an automatic temperature control system, an anti-ultraviolet photic film, a sound reinforcement system and a sealing shell, wherein the liquid crystal television driving plate is connected with the automatic brightness regulating system by an RS 232 interface for controlling the brightness of the liquid crystal television; an audio interface on the liquid crystal television driving plate is connected with the sound reinforcement system; the automatic temperature control system is used for transmitting temperature data to the liquid crystal television driving plate for controlling the work of a blower; and the anti-ultraviolet photic film is bonded to high photic glass in the front of the sealing shell and forms a closed temperature preserving space of an ultra bright television together with the sealing shell. The liquid crystal television system is widely applied to places such as walking streets, bus advertising kiosks, leisure squares, street signposts, ships, cars and the like.

The invention discloses a sound amplifying system with a classroom speech intelligibility measuring function. The sound amplifying system comprises a sound amplifying box, microphones, an audio host and the like; wherein the audio host adopts an improved STI indirect method to measure a speech transmission index; the sound amplifying box is used for amplifying lecture sounds of teachers and playing test excitation signals; the microphones are arranged at various measurement points in a classroom and used for receiving test signals and transmitting the received audio signals to the audio host;and a digital processing unit in the audio host processes the input digital signals, calculates the digital signals to obtain the speech transmission index (STI), and obtains the speech intelligibility of listening to the class at each measurement point according to the value of the STI. The method has the advantages of high precision, small error, real-time measurement, simultaneous measurement of a plurality of positions and the like, can eliminate measurement errors caused by individual voice differences of speakers, and can more accurately reflect the speech intelligibility of a real class.

The invention belongs to the technical field of public broadcast amplification, in particular to a multi-channel sound amplification system for adjusting the volume of a loudspeaker. The system comprises a plurality of sound amplification subsystems, each sound amplification subsystem corresponds to one channel, and each sound amplification subsystem performs echo cancellation by using the acquired sound signals and the sound source signals in all channels to obtain error signals. According to the obtained error signals, the noise sound pressure level of the current environment is obtained, the volume value of the loudspeaker in the sound amplification subsystem is calculated in combination with a mapping function, and the corresponding volume gain value of the loudspeaker at the current moment is obtained. By multiplying the sound signal to be played picked up from the sound amplification subsystem by the obtained loudspeaker volume gain value at the current moment, the processed sound source signal is obtained, and is played and amplified by the loudspeaker in the sound amplification subsystem.

The invention provides a loudspeaker array sound reinforcement system generating a uniform sound field, comprising: an input signal processing module (1), a filtering module (2), a multi-channel power amplifier (3) and a loudspeaker array (4); the filtering module (2) comprising: a parameter setting unit (201), a filter bank generating unit (202) and a filter bank processing unit (203); the filter bank generating unit (202) is configured to set The parameter design optimization problem set by unit (201): in the actual sound field energy efficiency is greater than or equal to J c Under the constraint condition, find a vector solution, make the difference of the actual sound field distribution calculated and target sound field distribution minimum; This vector solution is complex number weight coefficient vector; According to described complex number weight coefficient vector, obtain described filter bank processing unit ( 203) the time-domain impulse responses of the L filters. The sound reinforcement system of the invention can form a uniformly distributed sound field in the target area.

The invention discloses a sound amplification method of a sound amplification system. The sound amplification system comprises a sound amplification host, at least one near-talk voice acquisition device and at least one far-talk voice acquisition device, wherein the near-talk voice acquisition device and the far-talk voice acquisition device are respectively connected with the sound amplificationhost, and the distance between the near-talk voice acquisition device and a sound source is smaller than the distance between the far-talk voice acquisition device and the sound source. The sound amplification method of the sound amplification system comprises the following steps: carrying out audio pre-processing on the first audio signal when a first audio signal acquired by the near-talk voiceacquisition device and a second audio signal acquired by the far-talk voice acquisition device are received; and carrying out sound amplification playing of the first audio signal subjected after theaudio pre-processing. The invention also discloses the sound amplification system and a computer readable storage medium. According to the invention, the tone quality of the audio signal played through sound amplification is ensured.

This invention relates to 3D8-F; it is stereophony audio amplifier and playing (replaying) mode. It except the base audio amplifier system the left, middle, right three main channels, besides these increase six environment effective sound channels, include left, middle, right effective sounds, the back environment effective sounds, stage environment effective sounds, stage back environment effective sounds, ceiling environment effective sounds. The three main audio amplifier channels include the sub-low frequency channel, to finish the basic language and music audio amplifier and playing (replaying). The six environment effective sounds channels use to help finish the play, opera, stage play, essay and so on program perform and playing (replaying) various environment effective sounds.

The invention discloses a method for arranging a loudspeaker of an acoustic amplification system in a long space, aiming at a requirement for the non-uniformity of a sound field. According to the size and the boundary conditions of the long space and the requirement for the non-uniformity of the sound field, the method comprises the steps of: providing transverse and longitudinal arrangement numbers or intervals in the long space through acoustic attenuation curves of the loudspeaker at different positions in the long space. The method is adaptable to long spaces with two vertical boundaries being perpendicular to the ground surface and having consistent boundary conditions.