35 results about "Equalization (audio)" patented technology

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.

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.

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.

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 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.

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.

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.

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.

An audio device including at least one acoustic transducer and a controller is provided. The controller is configured to (1) render microphone calibration audio via the at least one acoustic transducer; (2) wirelessly receive a microphone calibration data set corresponding to the microphone calibration audio; (3) render sound optimization audio via the at least one acoustic transducer; (4) wirelessly receive a sound optimization data set corresponding to the sound optimization audio at a first listening location; (5) generate an equalization parameter based on the sound optimization data set and the microphone calibration data set; (6) generate an optimized output audio signal based on an output audio data set and the equalization parameter; and (7) render the optimized output audio via the at least one acoustic transducer based on the optimized output audio signal.

An audio device including at least one acoustic transducer and a controller is provided. The controller is configured to (1) render microphone calibration audio via the at least one acoustic transducer; (2) wirelessly receive a microphone calibration data set corresponding to the microphone calibration audio; (3) render sound optimization audio via the at least one acoustic transducer; (4) wirelessly receive a sound optimization data set corresponding to the sound optimization audio at a first listening location; (5) generate an equalization parameter based on the sound optimization data set and the microphone calibration data set; (6) generate an optimized output audio signal based on an output audio data set and the equalization parameter; and (7) render the optimized output audio via the at least one acoustic transducer based on the optimized output audio signal.

A system is provided. The system includes an analyzer for determining a plurality of binaural room impulse responses, and a loudspeaker signal generator for generating at least two loudspeaker signals depending on the plurality of binaural room impulse responses and depending on the audio source signal of at least one audio source. The analyzer is configured to determine the plurality of the binaural room impulse responses such that each of the plurality of the binaural room impulse responses considers an effect that results from a headphone being worn by a user.

The present disclosure is based on dynamic equalization during loudness equalization and DRC of encoded audio metadata. The invention provides for dynamic loudness equalization of received audio content in a playback system using metadata including a momentary loudness value for the audio content. A playback level is derived from a user volume setting of the playback system and compared to a mix level assigned to the audio content. A parameter is calculated that defines an equalization filter to filter the audio content before driving a loudspeaker with the filtered audio content, the parameter being based on the momentary loudness value and the comparison of the playback level to the assigned mix level. Other embodiments are described and claimed.

The invention discloses an audio processing method, apparatus and device, and a computer readable storage medium. The method comprises the steps of obtaining loudspeaker data of a plurality of loudspeakers and first distance information between each loudspeaker and a listening position; determining audio equalization data based on the loudspeaker data; determining crosstalk cancellation data based on the first distance information and second distance information between the plurality of speakers; and processing the to-be-processed audio data based on the audio equalization data and the crosstalk elimination data to obtain target audio data. By adopting the method provided by the invention, the acoustic image range of the loudspeaker can be expanded, and the influence of acoustic defects in a playback environment on the playback effect can be reduced.

The application discloses a kind of efficient parameter equalizer curve drawing method and processing terminal, method includes: forming N initial parameter equalization audio curve, with sampling frequency as abscissa, audio gain as ordinate;For each initial parameter equalization audio curve, sampling frequency is selected as abscissa according to corresponding interval size, the audio gain corresponding to each abscissa is calculated, to generate N final parameter equalization audio curve;The ordinate corresponding to each abscissa of each final parameter equalization audio curve is accumulated, to superimpose each final parameter equalization audio curve together, form a parameter equalization audio curve;The complete audio curve is loaded into the canvas Canvas of interface control, to obtain the audio curve of parameter equalizer.The application is less in data processing amount, interface loads the audio curve faster, responds faster, and interface jamming phenomenon does not appear.

A method of generating a personalized audio output response for a head-mounted device using a machine learning model. One or more images of a portion of a user's head are received, the one or more images including at least the user's ears. One or more features describing the user's ears are identified based on the one or more images. The features are input to a model, and the model is configured to determine an audio output response for the user based on the identified one or more features. The audio output response describes one or more acoustic parameters perceived by the user. A personalized audio output response is generated for the user based on the audio output response, the personalized audio output response configured to adjust one or more acoustic parameters of audio content provided to the user by the head-mounted device.

The application discloses a kind of end-side adaptive human voice beautification method and system for K song scene, belong to end-side audio signal processing and K song human voice beautification field, to solve the problem of fixed equalization, fixed reverberation or artificial parameter adjustment difficult to adapt to different sound line, input level, noise and sibilance state and host processing increase delay, the present application is analyzed by end-side acquisition audio, extracts human voice features, carries out credibility gate and scene stability, identifies human voice defects and maps and issues the constrained multi-section equalization parameter, realizes the technical effect of improving human voice beautification adaptability and end-side real-time processing stability.

This invention relates to the field of audio processing technology, and more particularly to a microphone modeling method, apparatus, device, and medium. The method involves acquiring raw audio data collected by a source microphone within a preset time period, performing convolution processing on the raw audio data to obtain first intermediate audio data, performing equalization modeling processing on the first intermediate audio data to obtain second intermediate audio data, performing transient modeling processing on the second intermediate audio data to obtain third intermediate audio data, performing nonlinear distortion modeling processing on the third intermediate audio data to obtain fourth intermediate audio data, performing proximity effect simulation processing on the fourth intermediate audio data to obtain fifth intermediate audio data, and generating a target microphone model based on the fifth intermediate audio data. Therefore, this application, through a multi-stage audio data processing flow, can accurately achieve microphone modeling, thereby generating a highly realistic target microphone model.

This invention discloses a method and system for enhancing audio effects in streaming media bitstreams, belonging to the field of computer technology. The method includes: receiving bitstream information transmitted by a streaming media service; acquiring preset audio effect information; loading a quantization parameter package based on the audio information through an inference model; transcoding the audio bitstream and then preprocessing the transcoded bitstream; determining an equalization feature sequence and a reverberation feature sequence based on the inference model; enhancing the bitstream with audio effects using an audio processing device based on the feature sequences; and playing the enhanced audio data through a playback device. The inference model includes an equalization feature inference layer and a reverberation feature inference layer. This invention adds audio effect enhancement functionality to audio transmission and playback, enabling dynamic adjustment of audio effects according to different songs, providing users with a better listening experience.

A method for generating an encoded bitstream from an audio program comprising a plurality of audio signals, the method comprising receiving, for each of the plurality of audio signals, information indicating a playback device with which the respective audio signal is associated, receiving, for each playback device, information indicating at least one of a delay, a gain, and an equalization curve associated with the respective playback device, determining, from the plurality of audio signals, a group of two or more related audio signals, applying one or more joint-coding tools to the two or more related audio signals of the group to obtain jointly-coded audio signals, and combining the jointly-coded audio signals, an indication of the playback devices with which the jointly-coded audio signals are associated, and indications of the delay and the gain associated with the respective playback devices with which the jointly-coded audio signals are associated, into an independent block of an encoded bitstream.

A non-transitory computer readable medium and an in-vehicle sound device relating to convert a vehicle audio into a surround sound and a sound system are provided. The non-transitory computer readable medium stores a computer-program product includes instructions for extracting features of an audio data for debugging based on a vehicle audio signal; performing classification according to the extracted features to separate the audio data into a plurality of single audio signals; mixing the plurality of separated single audio signals, and recording parameters of the current sound mixing algorithm as original sound mixing parameters; pre-configuring a plurality of spatial reverberation modes; debugging the original sound mixing parameters; and determining and saving optimized audio channel equalization parameters and optimized sound mixing parameters corresponding to the spatial reverberation modes as the multi-channel surround sound conversion parameters.

The invention relates to the technical field of audio and video transmission, and discloses an audio and video content adaptive transmission method and system, and the method comprises the steps: creating a video enhancement unit and a battery management unit based on the audio and video data flow and battery state data of security and protection monitoring equipment; performing video signal phase modulation on a monitoring camera in the security and protection monitoring equipment based on the video enhancement unit to obtain a phase configuration matrix; performing null phase adjustment on the battery management unit based on the battery state data and the phase configuration matrix to obtain an isolation phase matrix; residual normal batteries in the security and protection monitoring equipment are determined based on the isolation phase matrix, current balance parameters and transmission power distribution parameters are generated, the limitation of a traditional wired isolation mode is overcome, differentiated communication enhancement is provided for battery units in different positions and states, battery balance control of space perception is achieved, and the safety of the security and protection monitoring equipment is improved. The problem that a traditional unified control strategy cannot adapt to battery distribution differences is solved.

The invention discloses a method for optimizing steel wire recording digital audio quality, which comprises the following steps of: S1, measuring abrasion of a main shaft and a belt pulley of a steel wire recorder, and adjusting a tolerance to be within an original design range; s2, the magnetic head is cleaned through ultrasonic waves, and metal coating is conducted again; s3, controlling power supply voltage fluctuation by adding a voltage stabilizer; s4, analyzing the audio signal in combination with a spectrum analysis tool, separating distortion and noise components, and performing noise reduction processing by using a spectrum De-Noise technology and a dynamic threshold value; and S5, expanding a frequency range through a Spectral Recovery technology, and carrying out equalization compensation on the signal by using MusicRebalance. According to the method, mechanical structure optimization, magnetic head performance recovery, power system stability improvement and a digital audio processing technology are combined, so that the digital audio quality of steel wire recording is remarkably improved, the sound quality of recording is optimized, noise interference of equipment is reduced, and audio playback is clearer and more real.