81 results about "Equalization (audio)" patented technology

The embodiment of the invention discloses a sound field correction method and device and audio playing equipment. According to the embodiment of the invention, audio collection is carried out on the currently played content to obtain a current original audio file and a current echo cancellation audio file, and a current original frequency response curve corresponding to the current original audio file and a current echo cancellation frequency response curve corresponding to the current echo cancellation audio file are determined; and adjusting the current echo cancellation frequency response curve according to the reference equalization curve to obtain a target frequency response curve, determining a dynamic equalization curve according to the current original frequency response curve and the target frequency response curve, and further controlling audio signal output according to the dynamic equalization curve. Wherein the reference equilibrium curve is used for representing the difference between the reference original frequency response curve and the reference echo cancellation frequency response curve. Therefore, according to the embodiment of the invention, the audio playing equipment can realize sound field self-adaptive correction on the premise of not needing manual intervention, so that the user experience is improved.

The invention discloses a sound system with DSP sound effect enhancement processing, which belongs to the technical field of audio processing and comprises an audio input module, a data acquisition module, a DSP processing unit, an audio output module and a user interaction module. By acquiring characteristic data (such as audiogram and heart rate) of a hearing impaired person or a child in real time, a DSP processing unit dynamically calls a targeted algorithm (weak hearing frequency band compensation and voice rhythm adjustment), and a multi-frequency band equalization and noise suppression technology is combined, so that an optimized audio signal adaptive to user requirements is output. The problem that a traditional sound system cannot adapt to special crowds in a personalized mode is solved, the sound effect definition and user experience are remarkably improved, and the system is suitable for education, medical treatment and family scenes.

The invention relates to the technical field of wearable equipment and intelligent audio processing, in particular to an intelligent control method for TWS earphone heart rate detection and earphone function adaptive adjustment, which comprises the following steps: S1, synchronously acquiring an original heart rate signal, a contact pressure signal and a motion acceleration signal; s2, performing real-time baseline correction on the original heart rate signal; s3, the motion acceleration signals are fused to calculate motion interference weight factors, and a multi-modal physiological feature set is generated; s4, outputting a multi-dimensional physiological state vector; s5, matching a predefined adjustment strategy library according to the multi-dimensional physiological state vector; and S6, analyzing the combined control instruction set. According to the TWS earphone, physiological signals are accurately collected through the ear canal composite sensor array, high-precision correction of heart rate detection is achieved in combination with dynamic compensation, feature extraction and physiological state classification, noise reduction, audio equalization and ear canal pressure adjustment parameters are intelligently matched based on an adjustment strategy library, and the TWS earphone can adapt to the physiological state of a user.

Methods, apparatus, systems and articles of manufacture are disclosed to determine audio quality. Example apparatus disclosed herein include an equalization (EQ) model query generator to generate a query to a neural network, the query including a representation of a sample of an audio signal. Example apparatus disclosed herein also include an EQ analyzer to access a plurality of equalization settings determined by the neural network based on the query; and compare the equalization settings to an equalization threshold to determine if the audio signal is to be removed from subsequent processing.

Embodiments are provided for receiving a request to output audio at a first speaker and a second speaker of an electronic device, determining that the electronic device is oriented in a portrait orientation or a landscape orientation, identifying, based on the determined orientation, a first equalization setting for the first speaker and a second equalization setting for the second speaker, providing, for output at the first speaker, a first audio signal with the first equalization setting, and providing, for output at the second speaker, a second audio signal with the second equalization setting.

Systems and methods for equalizing audio during a collaboration session in a heterogenous computing platform are described. In an illustrative, non-limiting embodiment, an Information Handling System (IHS) may include: a heterogeneous computing platform comprising a plurality of devices, and a memory coupled to the heterogeneous computing platform, where the memory comprises a plurality of sets of firmware instructions, where each set of firmware instructions, upon execution by a respective device, enables the respective device to provide a corresponding firmware service, and where at least one of the plurality of devices operates as an orchestrator configured to: receive a policy from an Information Technology Decision Maker (ITDM) or Original Equipment Manufacturer (OEM), and select an audio equalization setting usable during a collaboration session based, at least in part, upon the policy.

The invention discloses an earphone space audio positioning method integrated with an MEMS sensor and a control system, and relates to the field of earphone space audio positioning. A temperature compensation MEMS sensor group and a microphone array are used for collecting motion data and sound wave signals, and signals are enhanced through orthogonal wavelet denoising and beam forming; the attitude is solved by adopting double-extended Kalman filtering and fusing data, and fuzzy logic is introduced to evaluate the credibility of the sensor; predicting a sound field by using a CNN model, and reconstructing the sound field in combination with an adaptive wave field synthesis technology; and based on auditory masking effect optimization balance, the influence of altitude on sound velocity is compensated. Through an ultra-precision sensor and an advanced algorithm, the positioning precision and reliability of earphone space audio are improved, and delay and power consumption are reduced; accurate sound source positioning, personalized sound field adaptation and cross-modal immersion experience in a complex scene are realized, and the audio experience of a user is improved.

A method performed by an audio system that includes a headset for playing back audio, the method includes receiving an audio signal having audio content, determining a characteristic of the audio content, determining a parameter of the headset, producing a filtered audio signal by applying an equalization (EQ) filter to the audio signal based on the characteristic of the audio content and the parameter of the headset, and driving one or more speakers of the headset with the filtered audio signal.

The invention relates to an automatic debugging method, device and equipment for a multimedia sound console and a medium, and the method comprises the steps: obtaining sound field space data, matching a pre-cached basic cloud template according to the sound field space data, obtaining an input audio signal of multiple sound channels, and transmitting the input audio signal to the multimedia sound console; determining a spectrum characteristic and a dynamic range characteristic of the input audio signal; carrying out equalization analysis on the spectrum features, and determining a corresponding equalization parameter set; determining a corresponding gain parameter set according to the dynamic range characteristics; according to the reverberation parameter, the equalization parameter set, the gain parameter set and the filtering parameter set, a corresponding local cache template is established, a cloud template matching technology based on sound field space data is introduced, the limitation that traditional tuning depends on fixed parameters or manual adjustment is overcome, the automation degree of tuning is improved, manual intervention is reduced, and the tuning efficiency is improved. And meanwhile, the stability of audio output quality is ensured.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

The invention discloses a self-adaptive sound equipment equalization calibration method. The method comprises the following steps: acquiring an audio signal of an audio system and carrying out short-time Fourier transform on the audio signal to obtain a frequency response curve; calculating a target level according to the frequency response curve; carrying out smoothing processing on the acquired frequency response curve; whether a peak value larger than a threshold value exists in the smoothed frequency response curve or not is judged, the influence of a current peak filter on frequency response is simulated through a Gaussian function according to peak filter parameters, the influence is superposed on the smoothed frequency response curve, and a new adjusted frequency response curve is obtained; if the number of iterations is greater than or equal to a preset value, ending iteration, outputting a peak filter parameter, and configuring a peak filter by adopting the peak filter parameter; and inputting the audio signal into a plurality of configured peak filters to obtain an audio signal after equalization calibration. The sound equipment equalization calibration efficiency is high.

A system (400) for providing an audio profile for streaming video, includes an audio profile subsystem (433) that receives audio mix data and synchronizes audio data in an audio profile. A streaming service video streaming server (420) receives video data, uncompressed audio data synchronized with audio tracks, and the audio profile. An audio profile server (430) is configured to receive the audio profile and dynamic range compression descriptors to produce audio profile metadata. An audio reproduction system (410) receives audio profile metadata and listening environment parameters to adjust audio parameters based on the audio rendering system characteristics and a listening environment of the audio rendering system. The audio parameters include one or more of equalization, a filter, a rendering delay, and volume.

Various aspects include wearable audio devices wearable audio devices with a control platform for managing external device interaction. In some particular aspects, a wearable audio device includes: at least one electro-acoustic transducer; at least one processor; and memory including instructions executable by the at least one processor, wherein the instructions are configured to apply a first equalization (EQ) setting to first audio provided by a first audio source, the first audio to be played using the at least one electro-acoustic transducer, and apply a second EQ setting different from the first EQ setting to second audio provided by a second audio source, the second audio source being different from the first audio source, the second audio to be played using the at least one electro-acoustic transducer.

The invention relates to the technical field of audio signal processing, in particular to an audio equalization adjustment method and device and a storage medium. The audio equalization adjustment method comprises the following steps: acquiring a physiological movement signal and an environment audio signal of a user; respectively processing the physiological motion signal and the environment audio signal to obtain physiological motion features and environment noise features; performing fusion decision based on the physiological motion features and the environmental noise features to generate a target equilibrium parameter set; and based on the target equalization parameter set, performing equalization adjustment on the audio output signal of the near-to-eye display device. According to the method, the auditory experience of the near-to-eye display equipment in the motion scene can be intelligently optimized.

1. The name of the design product: the main body of an audio processor. 2. The use of the design product: the main body as a protective shell for an audio processor, which is used as a whole to process audio signals, such as equalization, compression, reverb, delay, etc., to improve sound quality or achieve specific effects. 3. The design points of the design product: in shape. 4. The picture or photo that best shows the design points: perspective view 1.

The invention discloses a model training method and device, electronic equipment and a storage medium, and belongs to the technical field of audio processing, and the method comprises the steps: inputting a first audio frame parameter related to a first audio frame in a first audio signal into an audio equalization model for audio equalization processing, and outputting the gain of the first audio frame; adjusting the spectrum amplitude of the first audio frame by using the gain of the first audio frame to obtain an adjusted first audio frame; determining a first reward value based on the adjusted first audio frame; and on the basis of the first audio frame parameter, the gain of the first audio frame, the first reward value and a second audio frame parameter related to the second audio frame, updating the model parameter of the audio equalization model to obtain an updated audio equalization model.

The invention relates to the technical field of sound field control, in particular to a sound equipment sound field low-frequency increasing and decreasing method and system based on volume amplitude recognition, and the method comprises the following steps: collecting an original audio stream, and extracting instantaneous amplitude features; calculating a low-frequency compensation coefficient according to the nonlinear auditory model; constructing a dynamic safety threshold and correcting a coefficient in combination with the physical limit of the equipment; and adjusting the low-frequency component based on the corrected parameter, and generating an optimized sound field signal. According to the invention, by establishing a dynamic mapping and constraint mechanism between the instantaneous volume amplitude and the low-frequency gain, the technical defects that the low-frequency hearing feeling of a traditional sound system is deficient under the low volume and overload distortion is extremely easy to occur under the high volume are thoroughly solved, and under the premise of strictly guaranteeing the physical safe operation of loudspeaker hardware, the sound quality is greatly improved. And full-dynamic-range sound field adaptive equalization and restoration conforming to human ear psychological acoustic characteristics are realized.

Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization based on variant selection. An example apparatus to equalize audio includes at least one memory, machine readable instructions, and processor circuitry to at least one of instantiate or execute the machine readable instructions to train a neural network model to apply a first audio equalization profile to first audio associated with a first variant of media, and apply a second audio equalization profile to second audio associated with a second variant of media. The processor circuitry is to at least one of instantiate or execute the machine readable instructions to at least one of dispatch or execute the neural network model.

The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

The invention discloses an audio equalization method and device, a storage medium and a program product, and belongs to the technical field of audio. The method comprises the steps of obtaining a frequency spectrum of a target audio and a frequency spectrum of a reference audio, generating an EQ curve representing a frequency spectrum difference between the target audio and the reference audio, determining a statistical characteristic of the frequency spectrum difference, determining a compression parameter for performing range compression on the frequency spectrum difference based on the statistical characteristic, and performing range compression on the frequency spectrum difference based on the compression parameter and a soft knee compression algorithm. And performing optimization processing on the EQ curve to obtain an optimized EQ curve, and performing equalization processing on the audio signal of the target audio based on the optimized EQ curve to obtain an equalized audio signal of the target audio. Through soft knee compression processing, the spectrum distortion possibility of the balanced audio signal is reduced.

A 3D sound spatializer provides delay-compensated HRTF interpolation techniques, efficient cross-fading between current and delayed HRTF filter results, and per-object equalization and stabilization, to mitigate artifacts caused by interpolation between HRTF filters, the use of time-varying HRTF filters, and spectral coloration due to loudspeaker playback including acoustic crosstalk.

This invention provides a dynamic optimization method for noise suppression and frequency response equalization of an audio system in complex sound field environments. The method includes acquiring spatial acoustic signals, auxiliary sensing signals, and system reference signals; constructing a feature fusion dataset based on a physical-acoustic coupling mapping mechanism; performing spatiotemporal modeling of the current sound field environment; and generating a sound field state vector and a sound field complexity index. Based on the multimodal feature fusion dataset and the sound field state vector, a frequency domain energy residual surrogate model is introduced as a physical embedded constraint to perform feature-level bidirectional joint solution for noise suppression and dynamic frequency response equalization, generating an initial filter coefficient set. This invention, through soft decision-making and convex set projection, ensures that the final output filter coefficients simultaneously satisfy auditory masking protection and physical frequency response compensation, effectively avoiding the conflict between equalization amplification of background noise and noise reduction creating spectral holes.

A system that incorporates the subject disclosure may include, for example, a gaming system that cooperates with a graphical user interface to enable user modification and enhancement of one or more audio streams associated with the gaming system. In embodiments, the audio streams may include a game audio stream, a chat audio stream of conversation among players of a video game, and a microphone audio stream of a player of the video game. Additional embodiments are disclosed.

An earpiece includes a portion that seals the earpiece in the user's ear, preventing or reducing sounds from an external environment (e.g., the outside world) from entering the user's ear canal and being heard by the user. The earpiece also includes a duct from the user's ear canal to the external environment, which may be open or closed. Whether the duct is open or closed is automatically determined, such as based at least in part on audio measurements at a microphone of the earpiece. When the duct is open, sounds can pass from the external environment to the user's ear canal, active noise cancellation (ANC) is disabled, and an appropriate equalization (EQ) setting is used. When the duct is closed, sounds from the external environment are prevented from passing through to the user's ear canal (or are reduced), ANC is enabled, and an appropriate EQ setting is used.

The invention discloses a dynamic gain equalization method, device, equipment and product for sound equipment, and relates to the technical field of sound processing. The method comprises the following steps: performing empirical mode decomposition processing on each field sound signal from a microphone array to obtain corresponding N intrinsic mode function components, searching at least one adjacent frequency point group according to N component center frequency points of each signal, and searching at least one adjacent frequency point group according to each adjacent frequency point group to obtain N component center frequency points of each signal; the method comprises the following steps of: determining a corresponding sound source position according to a set of intrinsic mode function components, performing dynamic gain equalization processing on each intrinsic mode function component in each set when the sound source position is judged to be in a layout area of a loudspeaker to obtain a corresponding processed intrinsic mode function component, and finally reconstructing based on a processing result to obtain a processed sound signal. And the sound signals are transmitted to the loudspeaker for on-site playing, so that a wave trap is not needed, howling phenomenon suppression on the sound signals not from the loudspeaker can be avoided, and the aim of suppressing the howling phenomenon is fulfilled.

The embodiment of the application discloses an audio equalization method and device, a storage medium and an electronic device, and relates to the field of audio processing. The embodiment of the application introduces the extension bit result data Q[n] and Q[n-1] output by the quantizer, carries out multiplication operation with specific coefficients, carries out summation and inversion, generates compensation data Q3, ingeniously uses the redundant information in the quantization process, accurately compensates the quantization error, and thus significantly improves the fidelity and overall sound quality of the audio signal. The introduction of the compensation data Q3 makes the audio signal output by the quantizer closer to the original signal, reduces the distortion and noise caused by quantization, and brings a more pure and natural auditory experience to the user.

The invention discloses a balance adjustment method and system for a sound console volume controller, and belongs to the field of sound console control. The method comprises the steps that audio signals and environment noise are collected and preprocessed; audio state perception and noise identification are carried out by using a deep learning model combining a time sequence convolutional neural network and a long short-term memory network; a volume control and frequency equalization dual-channel parallel computing architecture is adopted, and controlled variables are generated based on a first-order active-disturbance-rejection control algorithm and a second-order active-disturbance-rejection control algorithm; using an improved particle swarm optimization algorithm to carry out collaborative optimization on the dual-channel control parameters; and finally synthesizing and outputting a high-quality audio signal. The system correspondingly comprises an audio acquisition module, a deep learning reasoning module, a dual-channel control module, a parameter optimization module and the like, and can be integrated with a hardware acceleration unit. The method effectively solves the problem of coupling of volume and frequency balance control, and has the advantages of high environmental noise adaptive capacity, intelligent parameter adjustment, high processing real-time performance and the like.

Generally disclosed herein is a mechanism to increase the perceived sound field width and depth in an automotive environment without causing excessive or unnatural reverberation. The audio ambiance may be added using a cross-analysis of a vehicle cabin and a typical room's sound decay characteristics. The late reverb insertion and extraction time may be determined from the target room response and the vehicle cabin response, and the late reverberation transfer function may be computed. Once the late reverberation transfer function is obtained, the combined cabin transfer function can be measured, and a scale factor for energy equalization derived. The total energy of the sound within the vehicle cabin may be normalized and the reverberation frequency spectrum may be equalized such that the late reverberation within the vehicle cabin is synthesized with the desired energy decay of the late reverberation of the sound in a room.

Example techniques described herein relate to a software application that is configured to operate as an add-on software component to audio-playback software on a playback device of a media playback system. An example implementation may involve adding the multiple audio tracks to the playback queue, and before playing back a first audio track, enabling a first add-on component to audio-playback software. The first add-on component corresponds to a first remote source and enabling this component causes a first modification to an equalization of the playback device. The example implementation may also involve before playing back a second audio track, enabling a second add-on component to the audio-playback software. The second add-on component corresponds to the second remote source and enabling this component causes a second modification to the equalization. Enabling the second add-on component disables the first add-on component if enabled.

The invention relates to the field of open earphones, in particular to a sound balancing system of an open earphone, which comprises a balancing system, the balancing system comprises a sound signal processing unit, a user hearing characteristic acquisition unit, an environmental noise acquisition unit, a sound balancing control unit, a multi-mode switching module and an earphone unit, the sound signal processing unit receives an audio signal through a receiving module, and collects hearing characteristics, environmental noise and sound signals of a user in real time through the arrangement of the sound signal processing unit, the user hearing characteristic collection unit, the environmental noise collection unit, the sound balance control unit, the multi-mode switching module and the earphone unit. According to the technical scheme, the volume, the tone, the timbre and other parameters can be automatically adjusted, the personalized and intelligent sound balancing effect is achieved, the tone quality parameters are automatically adjusted according to the hearing characteristics of the user and the environmental conditions, the personalized and intelligent sound balancing effect is achieved, and the hearing experience of the user is improved.