29 results about "Sound reinforcement system" patented technology

The embodiment of the invention is applied to the field of artificial intelligence, and provides an audio processing method, a de-howling model training method and electronic equipment, and the method comprises the steps: obtaining a to-be-processed audio, the to-be-processed audio being obtained according to an audio collected by a microphone in a sound reinforcement system; de-noising processing is performed on the plurality of first audios in sequence by using a de-howling model to obtain a plurality of second audios, the first audio corresponding to the first de-noising processing is the audio to be processed, and the first audios corresponding to other de-noising processing except the first de-noising processing are the second audios obtained by the previous de-noising processing. The second audio obtained by the last denoising processing is a squeal-removed audio, the squeal-removed model is a neural network model obtained by training, and a loudspeaker in the sound reinforcement system is used for converting the squeal-removed audio into a sound signal. Based on the technical method provided by the invention, the howling in the audio to be processed can be accurately removed.

The invention discloses a conference sound reinforcement system based on an AI intelligent algorithm and 360-degree omni-directional noise reduction, and relates to the technical field of audio signal processing, the conference sound reinforcement system comprises a conference management center, the conference management center is in communication connection with the following modules: a multi-sound-source sensing module used for capturing 360-degree omni-directional sound field information in a conference environment and constructing a sound source space distribution model; according to the invention, the omnidirectional microphone array unit covers all directions of a conference space, synchronously collects audio data streams, eliminates the limitation of a conventional unidirectional microphone, combines the sound source positioning space mapping unit, constructs a sound source space distribution model based on the time difference of arrival and the phase difference through a deep learning model, and improves the sound source positioning accuracy. The azimuth angle, pitch angle and distance parameters of the sound source are accurately analyzed, the position of the sound source is mapped to a virtual space coordinate system, a dynamically updated 3D sound source distribution diagram is generated, the position change and intensity distribution of the sound source are reflected in real time, and the positioning precision in a complex acoustic environment is remarkably improved.

The invention relates to a howling suppression method, in particular to a howling suppression method based on variable step size adaptive filtering and trapped wave complementation. The invention aims to solve the problem of howling suppression caused by sound feedback in an audio sound reinforcement system. According to the method, a new error estimation and adjustment strategy is introduced, the updating process of the adaptive filter is optimized, the suppression precision is remarkably improved, and howling detection blind compensation is carried out in combination with the notch filter. The algorithm can effectively identify and suppress the howling component in the audio signal in real time, reduce the system delay, and maintain the definition and stability of the tone quality. The invention belongs to the technical field of voice signal processing.

The present invention proposes an evaluation method for evaluating the sound reinforcement performance of a sound reinforcement system in a target building space. First, a corresponding spatial model of the target building space to be evaluated is established according to its internal form; then, the spatial model of the target building space established in step 1 is partitioned according to the sound reinforcement system layout plan of the target building space; then, the spatial sound reinforcement dissipation coefficient of each partitioned space is calculated; finally, the spatial sound reinforcement dissipation coefficient β of the sound reinforcement system layout plan of the entire target building space is calculated; the smaller the obtained β, the smaller the impact of the sound reinforcement on other spaces, and the better the sound reinforcement performance of the sound reinforcement system layout plan of the target building space.

1. The name of the design product: network linear active sound column. 2. The use of the design product: for public broadcasting sound amplification system. 3. The design points of the design product: in shape. 4. The picture or photo that best indicates the design points: perspective view.

An acoustic feedback control method for a low-time-delay acoustic reinforcement system comprises the following steps: constructing an acoustic reinforcement system model, establishing a closed-loop transfer function, and determining a system stability condition; constructing an acoustic feedback path, generating an open-loop training data set, and simulating an acoustic feedback signal in a critical unstable state; extracting time-frequency characteristics of the input signal, and designing a complex spectral mapping target of the neural network; constructing a full-sub-band packet long-short term memory network which comprises an encoder, a full-band and sub-band processing module and a decoder, training the network by adopting a mixed amplitude spectrum and complex spectrum loss function, and optimizing a signal reconstruction strategy to reduce time delay; finely adjusting the network in a simulated closed-loop system to adapt to an actual acoustic environment and nonlinear distortion, and limiting the amplitude of an output signal through hard tailoring; and deploying the fine-tuned model to a real sound reinforcement system, and cooperatively inhibiting sound feedback with a frequency shift method. According to the method, sound feedback can be effectively inhibited under the condition of relatively low time delay, the quality and intelligibility of voice are improved, and the maximum stable gain of a sound reinforcement system is improved.

The invention discloses an electronic conference file management method and system, and the method comprises the steps: starting a conference according to a conference agenda, and sequentially starting a conference unit corresponding to each spokesman; each conference unit acquires audio and video of a corresponding speaker through a pickup microphone and a camera and transmits the audio and the video to the conference host in real time, and the conference host transmits the audio to the sound amplification system for field sound amplification and transmits the video to the screen for displaying and playing; the audio is converted into a text in real time, timestamps are synchronously added into the audio and the video in combination with Beidou time service and network time service protocols, and then the audio, the video, the text and the timestamps are stored in real time to form all-media conference data; and the upper computer gathers the all-media conference data of each conference unit and performs classified storage to form an electronic conference file, so that a large amount of labor and time are saved, and the electronic conference file stored according to a preset rule is directly formed after the conference.

The application provides a method and system for visual monitoring of emergency conference system equipment operation, audio signals at multiple positions in a target conference room are collected; a mixed signal formed by the audio signals is retrieved, the mixed signal is separated into multiple audio channels to form a matrix; a buffer area is set for each audio channel; the order of audio data input and output is controlled, left and right channel audio data of each channel is stored in a corresponding slave buffer area, and the slave buffer area is recalled to a master buffer area; volume size analysis, echo detection, howling detection and bottom noise detection are performed on audio data of the master buffer area; and audio quality abnormal conditions are recorded and displayed. The application realizes detection of the running state of an audio processor and a public address system, facilitates monitoring of the running state of equipment by operation and maintenance personnel, and improves operation and maintenance efficiency.

A low-latency sound reinforcement system acoustic feedback control method, comprising: constructing a sound reinforcement system model, establishing a closed-loop transfer function, and determining the system stability condition; constructing an acoustic feedback path and generating an open-loop training data set to simulate the acoustic feedback signal under a critical unstable state; extracting the time-frequency features of the input signal and designing the complex spectrum mapping target of the neural network; constructing a full-subband grouping long-short term memory network, including an encoder, a full-band and sub-band processing module, and a decoder, training the network using a hybrid amplitude spectrum and complex spectrum loss function, optimizing the signal reconstruction strategy to reduce the latency; fine-tuning the network in the simulated closed-loop system, adapting to the actual acoustic environment and nonlinear distortion, and limiting the output signal amplitude through hard clipping; deploying the fine-tuned model to the real sound reinforcement system and cooperating with the frequency shift method to suppress acoustic feedback. The method can effectively suppress acoustic feedback at a lower latency, improve the quality and intelligibility of speech, and increase the maximum stable gain of the sound reinforcement system.

The invention relates to a multi-tone-area high-fidelity vehicle-mounted sound amplification system, which constructs a compartment acoustic fingerprint through an ultrasonic transmitter group, a microphone array and a pressure / environment sensor, solves equivalent point coordinates (x, y, z) and a sound absorption coefficient alpha in real time, and drives a compensation matrix to dynamically optimize the frequency response of a loudspeaker in combination with a passenger coefficient Occup and a temperature and humidity offset delta E. Performing ultrasonic scanning based on Gold code spread spectrum, and completing three-dimensional modeling within 1.8 seconds; a 4 * 3 compensation matrix is originally created to solve load / material / temperature and humidity disturbance, and the full-working-condition frequency response fluctuation is smaller than or equal to + / -1.5 dB; the full-duty low-frequency attenuation is improved by 300%, the sound image drift is smaller than or equal to + / -3 degrees, the voice definition MOS value reaches 4.1, and intergenerational crossing of the vehicle-mounted audio self-adaptive technology is represented.

The embodiment of the utility model provides an audio sound amplification system, and the system comprises a box body which is internally provided with at least one function module; the function module comprises at least one of a smart host module, a POE switch module, a sound box module, a control panel module, a Bluetooth module, a wire arrangement module and an extension socket module. A plurality of rollers are arranged at the bottom of the box body; and the box cover is detachably connected with the box body.

The application relates to a sound amplification system control method, an intelligent terminal and a storage medium, and belongs to the field of acoustic control. The method comprises the following steps: acquiring a horizontal coverage angle of a to-be-tested sound box of the sound amplification system; acquiring room information of a target room; acquiring an auditory distance between a listener in the target room and the to-be-tested sound box; calculating an installation quantity of the to-be-tested sound box based on the auditory distance, the room information and the horizontal coverage angle; measuring a reverberation time of the to-be-tested sound box in the target room based on a microphone of the sound amplification system; judging whether the reverberation time is located in a preset standard reverberation time range; and if not, issuing prompt information for increasing a sound absorption coefficient of sound absorption material in the target room. The application has the effect of effectively reducing the probability of poor sound amplification effect of the sound amplification system caused by personnel experience in installing sound compensation boxes.

1. Name of the product in this design: Digital Feedback Suppressor (DF-2). 2. Purpose of this design: Used in a sound reinforcement system to suppress howling and acoustic feedback phenomena of loudspeakers. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: a 3D model.

The invention discloses a wireless microphone sound amplification system based on a Hall switch. The wireless microphone sound amplification system comprises a wireless microphone transmitting device, a wireless microphone receiving device and a wireless microphone transmitting device, wherein the wireless microphone transmitting device comprises a microphone, a first wireless module, a first main control module and a magnet; the wireless microphone receiving device comprises a loudspeaker, a second wireless module, a second main control module and a Hall switch; and the pairing control system triggers pairing when the distance between the magnet and the Hall switch is less than or equal to 5cm, and the wireless microphone receiving device screens the wireless microphone transmitting device through a dynamic RSSI (Received Signal Strength Indicator) threshold value. According to the invention, the Hall switch triggering and the RSSI secondary screening are combined, and the sensing distance is less than or equal to 5cm, so that compared with a traditional RSSI-based pairing mode, the accuracy of pairing triggering is greatly improved, unnecessary misoperation is avoided, and the error connection rate is effectively reduced; and the standard operation of teachers is cultivated through a touch type pairing mode, so that the operation error rate is reduced, and equipment faults and inconvenience in use caused by misoperation are reduced.

The present application relates to the field of audio technology, especially to a hanging microphone audio amplification system, comprising: a hanging microphone for converting sound waves into electrical signals; an audio detection module for obtaining a first-level audio feature parameter, comprising a vibration sensor; an audio data processing module for outputting a second-level audio feature parameter by analyzing and calculating the first-level audio feature parameter; a central control module for initially adjusting the rotation speed of the hanging microphone according to the signal-to-noise ratio of the audio, or initially adjusting the maximum rotation angle of the hanging microphone according to the vibration frequency of the hanging microphone, and secondarily adjusting the rotation speed of the hanging microphone according to the variance of the fluctuation amplitude of the audio, and secondarily adjusting the maximum rotation angle of the hanging microphone according to the difference amount of the volume; the present application realizes the improvement of the collection effectiveness of the hanging microphone audio.

The invention provides a sound reinforcement anti-howling method and a sound reinforcement system, and relates to the technical field of audio playing. The sound reinforcement anti-howling method comprises the following steps: processing a sound signal by using a pre-amplification unit to generate a first signal, and controlling a howling detection unit to detect and receive the first signal and obtain an output signal corresponding to the first signal; if it is determined that a howling condition is met according to the output signal and the first signal, the center frequency of the trap filtering unit is adjusted to avoid howling, and the howling condition includes that the system gain exceeds a preset range and the gain change rate exceeds a preset change threshold. According to the embodiment of the invention, the howling can be accurately detected and the frequency related to the howling in the signal can be suppressed, so that complex sound field changes can be quickly adapted, the stability is good, and the application range is effectively expanded.

The embodiments of the present application provide a howling suppression method, device, audio and sound amplification system, and relate to the field of digital signal processing technology. The method includes: pre-processing the audio signal in the sound reinforcement system and converting the audio signal into the frequency domain; processing the converted audio signal frame by frame and frequency point by frequency point based on the filter coefficient of the frequency domain adaptive filter to obtain the corresponding output signal, and at the same time using the far-end audio signal in the current frame signal as the reference signal to update the filter coefficient for use in the next frame signal processing; converting all the obtained output signals into the time domain to obtain the target audio. In the embodiment of the present application, since the reference signal when the filter coefficient is updated is the far-end signal, it not only reduces the correlation between the filter input signal and the reference signal, but also effectively suppresses the feedback signal.

The present invention provides a delay measurement method and device for a sound reinforcement system, an electronic device, and a storage medium. The delay measurement method includes: playing a preset first audio signal for the first time by a loudspeaker, picking up a second audio signal by a pickup, and processing the second audio signal through a speech algorithm to obtain a third audio signal; obtaining an adjustment parameter of the sound reinforcement system according to signal processing of the second audio signal and the third audio signal; playing the first audio signal for the second time by the loudspeaker, and using the adjustment parameter to adjust the obtained third audio signal in real time; obtaining the sound reinforcement delay of the sound reinforcement system according to the self-excitation oscillation information of the third audio signal. Thereby, the accuracy and operability of the sound reinforcement system delay measurement can be improved, the error caused by manual intervention can be avoided, the cost can be saved, and the debugging efficiency can be improved.

The utility model relates to the field of sound amplification equipment, in particular to a sound control circuit and a sound amplification system, and the sound control circuit comprises a sound channel selection module, a resistor array, a sound output module, a main control module and an anti-interference module. The sound channel selection module is respectively connected with the main control module and the resistor array, and the resistor array is connected with the sound output module; the main control module is connected with the anti-interference module, the anti-interference module is connected with the sound output module, and the technical problem that sound interruption or mechanical noise is prone to occurring in an existing volume controller is solved.

The invention provides an automatic evaluation system and method for a vehicle-mounted sound reinforcement system. Efficient and objective whole vehicle acoustic performance evaluation is realized through binaural microphone multi-dimensional acquisition, pulse response analysis and a dynamic weighted scoring model. According to the system, a binaural microphone is fixed by a multi-dimensional bracket with a height and distance adjusting mechanism, and the head posture of a passenger is accurately simulated; a 20Hz-20kHz sweep frequency signal and an 18-24kHz near ultrasonic signal are driven in a time-sharing manner through a loudspeaker controller, and crosstalk is suppressed in combination with a time division multiplexing technology; six indexes such as frequency response flatness, reverberation time, total harmonic distortion and a speech transmission index are extracted based on an impulse response file, a centesimal system score is generated by adopting reverse normalization and dynamic weight adjustment, and the problems of spatial distortion, high-frequency detection blind areas, subjective and objective disjunction and the like of traditional single-point measurement are effectively solved.

The application provides an audio signal processing method for a multi-person conference scene of a distributed multi-microphone sound amplification system. The distributed multi-microphone sound amplification system comprises a sound pickup module, a sound amplification module, a networking module and an audio processor. The sound pickup module comprises at least two microphones arranged independently, and the microphones correspond to speakers one by one. The sound amplification module comprises at least one loudspeaker. The at least two microphones and the at least one loudspeaker are in communication connection with the networking module and the audio processor. The distributed multi-microphone sound amplification system automatically identifies a sound state through audio parameter analysis (signal energy intensity + human voice feature information), determines a processing strategy for different audio signals according to the sound state, ensures clear sound amplification of effective speech and suppression and weakening of invalid or whisper signals, and avoids multi-microphone noise superposition without manual intervention. Even if multiple people speak alternately, the dynamic optimization of the audio signal can be quickly and accurately completed, and the complex conference scene requirements can be adapted.

The invention relates to the field of dynamic howling suppression, in particular to a real-time dynamic howling suppression algorithm based on an FPGA (Field Programmable Gate Array), which comprises the following steps: S1, signal acquisition and conversion; s2, howling detection is carried out; s3, noise injection and path estimation; and S4, howling suppression and signal output. According to the invention, the FPGA chip is used as a digital signal processing core and is combined with the audio acquisition chip and the loudspeaker to form a hardware basic platform of a howling sound amplification system; the system adopts a digital audio signal with a 32KHZ sampling rate and 16-bit data precision to monitor the squeal phenomenon in real time, and when squeal occurs, it is verified that sound feedback path parameters are estimated within 160ms and squeal suppression is realized; according to the scheme, the advantages of low delay of the dynamic howling suppression algorithm and parallel processing of the FPGA are combined, and the real-time performance of the dynamic howling suppression algorithm is further improved.

Improved adaptive feedback cancellation may be used to improve performance of audio amplification systems, such as hearing assistance devices, sound reinforcement systems, telephony, and other acoustic amplification and reproduction systems. This adaptive feedback cancellation allows a significant increase in the maximum stable gain of the amplification system, such as by increasing gain while reducing or eliminating feedback. This improves the audibility provided by an audio amplification system. This may provide particular improvements for hearing assistance devices that include open fittings or otherwise have substantial acoustic leakage. This adaptive feedback cancellation provides additional protection from a dynamically changing acoustic leakage by continually updating itself to model the changes, thereby providing increased gain while reducing or eliminating feedback.

The invention discloses a report hall sound reinforcement system based on an SIS system principle, and relates to the technical field of control systems, and the system comprises a multichannel input interface, a main sound reinforcement array, an auxiliary loudspeaker array, an ultra-low frequency loudspeaker, a matrix processor, and a self-adaptive controller. The directional radiation design of the main sound amplification array is combined with the distributed auxiliary loudspeaker topology, so that the whole-field sound pressure uniformity is improved, and the problem of non-uniform sound field distribution of a traditional system is solved; a middle sound channel independent regulation and control technology optimizes music signal restoration quality while guaranteeing language definition through a dynamic frequency division point switching mechanism, solves a long-term pain point of compatibility of music and human voice in the industry, effectively inhibits crosstalk between sound channels through a phase correction algorithm of a matrix type audio processing module, improves sound positioning precision, and improves sound positioning accuracy. A sound positioning fuzzy phenomenon is eliminated; the ground coupling design of the ultralow-frequency module reduces low-frequency harmonic distortion, realizes high-efficiency transmission of low-frequency energy and avoids interference on medium-high-frequency sound image positioning.

The invention discloses a structure for compressing a high-pitch horn, and belongs to the field of electroacoustic transducers. The structure comprises a horn body, and is characterized in that the contour line of the inner wall of the horn is divided into two sections with different functions, namely a high-frequency control section close to a throat and a diffusion form section close to a mouth part. The high-frequency control section is composed of strong influence line segments with precisely limited geometric parameters and is specially used for controlling high-frequency acoustic performance; the diffusion form section is composed of weak influence line segments which can be flexibly adjusted and is mainly used for controlling the sound wave diffusion form. According to the design method, high-frequency performance control and physical size and diffusion design are decoupled, so that flat frequency response and low-distortion characteristics can still be realized under the condition that the axial height is limited. In addition, the bridge mounting structure is integrally formed outside the horn body, so that the horn body and the mid-bass unit can be integrally mounted conveniently. The invention is especially suitable for small and medium-sized professional sound amplification systems with strict requirements on volume.

The invention provides a voice enhancement and directional area rendering sound reinforcement system and method based on a microphone array, and the system comprises the microphone array, a voice signal processing device, a first loudspeaker array, and a second loudspeaker array. The microphone array is used for collecting sound signals in a target space; the voice signal processing device is used for determining a first voice signal of a first area and a second voice signal of a second area in a target space from the sound signals, and processing the first voice signal and the second voice signal to obtain a first audio output signal and a second audio output signal; the first loudspeaker array outputs according to the first audio output signal, the output bright area is arranged in the second area, and the output dark area is arranged in the first area; the second loudspeaker array is used for outputting according to the second audio output signal, a bright area output by the second loudspeaker array is arranged in the first area, and a dark area output by the second loudspeaker array is arranged in the second area. According to the invention, the isolation effect during inter-talking in the scene is improved, and the sound wave interference phenomenon is avoided.

The invention discloses a method, a device and equipment for testing the state of a hanging microphone sound amplification system, and the method comprises the steps: firstly setting the states of all sound boxes as non-mute states; then, the test audios in the test audio set are played in sequence through the sound box, audio collection is carried out through the hanging microphones, and test files corresponding to the test audios are generated; respectively judging whether the test file generated by each hanging microphone is consistent with the test audio or not; if yes, marking the microphone hanging state, and performing sound pressure level calculation based on the test file; judging whether each sound pressure level is lower than a preset value or not; if so, displaying that the sound pressure levels of the hanging microphone and the sound box are low. According to the method for testing the state of the hanging microphone sound reinforcement system, the connection state of the hanging microphone and the sound box in the sound reinforcement system can be tested based on the preset process, and the testing efficiency of the sound reinforcement system is improved. And moreover, not only can the connection state be tested, but also the sound pressure can be tested, so that the state test result of the hanging microphone sound amplification system is more accurate.

The invention provides a method and device for building a sound reinforcement system in a small place, and the method comprises the following steps: building an initial sound reinforcement system in a target scene based on an initial building scheme, and carrying out the acoustic collection in the target scene, and obtaining acoustic data; constructing a multi-dimensional acoustic matrix based on the acoustic data; inputting the multi-dimensional acoustic matrix into a preset feature processing model to obtain a one-dimensional first acoustic matrix; constructing a working characteristic matrix based on the arrangement position and the working parameter of the sound reinforcement facility in the initial construction scheme; constructing input features based on the working feature matrix and a first acoustic matrix, inputting the input features into a preset parameter adjustment model, outputting a feature adjustment matrix by the parameter adjustment model, and enabling the feature adjustment matrix to correspond to the working feature matrix, and adjusting the arrangement position and the working parameter of each sound reinforcement facility based on the feature adjustment matrix.

The application provides a small venue sound amplification system building method and device, the method comprises the following steps: based on the initial building scheme, an initial sound amplification system is built in the target scene, and acoustic data is obtained by performing acoustic acquisition in the target scene; a multi-dimensional acoustic matrix is constructed based on the acoustic data; the multi-dimensional acoustic matrix is input into a preset feature processing model to obtain a one-dimensional first acoustic matrix; a working feature matrix is constructed based on the arrangement position and working parameters of the sound amplification facilities in the initial building scheme; an input feature is constructed based on the working feature matrix and the first acoustic matrix, the input feature is input into a preset parameter adjustment model, the parameter adjustment model outputs a feature adjustment matrix, the feature adjustment matrix corresponds to the working feature matrix, and the arrangement position and working parameters of each sound amplification facility are adjusted based on the feature adjustment matrix.