65 results about "Audio filter" patented technology

The invention provides an audio processing device and method, an augmented reality device, equipment and a storage medium, and belongs to the technical field of virtual reality display and sensing. The audio processing device comprises a positioning device which is configured to track and position a user in a real space in real time, determine positioning information of the user and send the positioning information to a processor; the processor is configured to determine target positioning information of the user based on pre-established spatial information and the real-time positioning information of the user; determining association relationship information between the user and a virtual sound source in a virtual space based on pre-established space information and real-time target positioning information of the user; obtaining human body parameters of the user, and determining an audio filtering result matched with the user based on the association relationship information, the human body parameters and the audio frequency of the virtual sound source; and reconstructing the audio to be played based on the audio configured for the virtual sound source in advance and the audio filtering result.

A media system and a method of using the media system to reproduce spatial audio based on head-to-torso orientation, are described. The method includes determining a head-to-source orientation and a head-to-torso orientation based on head orientation data generated by a head tracking device. Determining the head-to-torso orientation includes determining torso movements based on movements of the head. The torso can be determined to move when the head movements meet a head movement condition, such as a predetermined angle of movement or pattern of movement. A binaural audio filter that is based on a head-related transfer function corresponding to both the head-to-source orientation and the head-to-torso orientation is applied to an audio input signal to generate an audio output signal. The audio output signal is played to accurately recreate spatial audio having sounds emitted to the user by a sound source. Other aspects are also described and claimed.

A system for synchronizing and spatializing media, comprising client device and server device. A method for operation includes: tracking the real-time spatial coordinates of both devices via wireless sensors; synchronizing media playback using a timestamp-based protocol that compensates for network latency and clock offset; and applying directional audio filters, such as Head-Related Transfer Functions (HRTF), on the client device to render audio appearing to originate from server device's physical location. The method is further characterized by compensating for acoustic propagation and signal processing delays to ensure accurate spatiotemporal alignment. The system may be enhanced by a multimodal Al configured to perform operations such as real-time voice-preserving translation and the generation of synchronized, spatialized haptic feedback based on semantic event markers.

The utility model provides an audio filtering mainboard. The audio filtering mainboard comprises an input module, an amplification module, a filtering module and an output module, the output circuit is connected with the modulator and the filtering module; the modulation circuit is used for amplifying the audio signal output by the input module and removing noise coupled by the circuit to obtain a relatively pure initial signal, and the initial signal processed by the modulation circuit is converted from an analog signal to a digital signal, so that distortion of the signal in the transmission process is reduced, and the transmission efficiency is improved. And the filtering module is used for filtering the adjusted modulation signal, so that the negative influence of noise and EMI (Electro-Magnetic Interference) on the tone quality is reduced, the audio quality is improved, and the electronic sound equipment obtains a good tone quality effect.

The invention discloses an audio somatosensory vibration system based on NTC temperature monitoring and application thereof, and belongs to the technical field of audio somatosensory vibration.The audio somatosensory vibration system comprises an audio vibrator, an audio access module and an audio control unit, the audio control unit is electrically connected with the audio vibrator and the audio access module, and the audio vibrator is electrically connected with the audio access module. An NTC temperature monitoring module is integrated in the audio oscillator, and the audio oscillator is electrically connected with an arranged vibration wave output module; the audio access module is electrically connected with an external audio source; the audio control unit is internally provided with an NTC temperature discrimination module and an audio filtering module in an integrated manner, and the audio filtering module carries out low-pass filtering processing on audio signals accessed by the audio control unit and extracts low-frequency components. According to the invention, multi-scene application can be realized, and the equipment is ensured to operate in a safe temperature range, so that the service life of the equipment is prolonged and the user experience is improved.

Approaches provided herein can provide for adjustments to audio data. In particular, an audio resampling process can be performed wherein the correct audio frame size is determined. A “target” audio stream duration, as may be indicated by a time-to-sample atom (STTS) atom, can be used to apply one or more pitch-invariant, time-stretching audio filters to the audio data as that data is decoded during a media transcode process. A resulting output media metadata can specify accurate, constant audio frame durations, and the audio stream duration can match the corresponding video stream duration in a media file.

A multi-input, multi-output audio processing is implemented as a linear system for use in an audio filter bank to convert a set of frequency-domain input audio signals into a set of frequency-domain output audio signals. The transfer function from one input to one output is defined as a frequency-dependent gain function. In some implementations, the transfer function includes a direct component that is substantially defined as a frequency-dependent gain and one or more decorrelating components having a frequency-varying group phase response. The transfer function is formed by a set of sub-band functions, where each sub-band function is formed by a set of corresponding component transfer functions that include the direct component and one or more decorrelating components.

The present invention discloses a smart noise reduction device including a control device; an audio waveform pattern recognizer coupled to the control device for identifying an audio mixed signal including a regularity signal and a non-regularity signal; an audio waveform pattern database coupled to the control device, including at least one audio type, each having a plurality of preset second regularity signals; and an audio filter coupled to the control device to obtain the regularity signal.

To provide a device and a method for spatial audio-filtering in a spatial audio capture.SOLUTION: A device includes means configured to acquire a plurality of audio signals from a plurality of microphones, determine a first sound source direction parameter and a first sound source energy parameter in one or more frequency bands of the plurality of audio signals on the basis of processing of the plurality of audio signals, determine a second sound source direction parameter and a second sound source energy parameter in one or more frequency bands of the plurality of audio signals on the basis of processing of the plurality of audio signals, acquire an area for defining a direction and / or a range for a filter, and generate a filter to be applied to a plurality of audio signals. The device generates a filter gain / attenuation parameter on the basis of an area associated with the first sound source direction parameter, the first sound source energy parameter, the second sound source direction parameter, and the second sound source energy parameter.SELECTED DRAWING: Figure 1

The present disclosure belongs to the technical field of virtual reality display and sensing. Provided are an audio processing apparatus and method, and an extended reality apparatus, a device and a storage medium. The audio processing apparatus comprises: a positioning device, which is configured to track and position a user in a real space in real time, determine positioning information of the user, and send the positioning information to a processor; and the processor, which is configured to determine target positioning information of the user on the basis of pre-built spatial information and real-time positioning information of the user, determine association relationship information between the user and a virtual sound source in a virtual space on the basis of the pre-built spatial information and real-time target positioning information of the user, acquire human body parameters of the user, determine, on the basis of the association relationship information, the human body parameters and an audio frequency of the virtual sound source, an audio filtering result matching the user, and on the basis of an audio that is configured for the virtual sound source in advance and the audio filtering result, rebuild an audio to be played.

A system and method for synchronizing and spatializing media between client device and server device. Client-side artificial intelligence application determines the position of server device by fusing data from at least two sources: visual data obtained from a camera and acoustic data obtained from a predetermined reference signal emitted by server device. This data fusion enables continuous spatial tracking, including during periods of visual occlusion. A timestamp-based protocol is used to synchronize media playback. Based on the tracked position, client device applies directional audio filters, such as Head-Related Impulse Response (HRIR), to render audio that appears to originate from server device's physical location. The method further compensates for acoustic propagation and signal processing delays to ensure accurate spatiotemporal alignment.

A computer-implemented method, when executed by data processing hardware, causes the data processing hardware to perform various operations. These operations include receiving an audio signal from a sensor array at a unified post-filter, converting the audio signal to a short time Fourier transform (STFT) domain via a conversion function, determining a speech presence probability based on the converted audio signal, and determining a noise smoothing factor based on the speech presence probability. The operations further include estimating, via the unified post-filter, a noise power spectral density based on the noise smoothing factor, estimating, via the unified post-filter, a steering vector of the desired source, and generating, via the unified post-filter, a directivity-based mask and a coherence-based mask. Operations further include generating a residual echo spectrum estimate based on the directivity-based mask and the coherence-based mask, and setting one or more spectrum shaping factors based on the residual echo spectrum estimate.

The application discloses a digital filtering reconstruction method for lossless audio, and relates to the field of audio filtering reconstruction. Firstly, the transient vibration intensity of the segmented audio frame is extracted in real time to obtain a transient index value, and the transient index value is mapped into a window asymmetry factor by using a nonlinear control curve. According to the factor, a front half cycle attenuation compression and a rear half cycle stretching are applied to a reference window function to generate a dynamic asymmetric window and further obtain dynamic filter coefficients. Finally, the audio is weighted and convoluted by using the dynamic coefficients, and a cross-fading splicing technology is combined to eliminate frame boundary artifacts caused by dynamic adjustment. In this way, dynamic reconstruction can be realized only by relying on time-domain point-by-point modulation of a single set of filters, and the storage of a large number of filter banks or complex offline conversion requirements are completely eliminated.

A listening system includes one or more remote signal sources that generate one or more electronic signals, which correspond to sound in an ambient environment. A first ear microphone detects a first combination of audio signals including ambient sound and one or more propagated audio signals, corresponding to the one or more electronic signals, received at a first ear of a listener. The system includes first ear playback device and a processing device to apply a first set of audio filters including an audio filter to process a respective electronic signal of the one or more electronic signals with a first error signal, which is based on an output of the first ear microphone, to generate a first output signal to the first ear playback device, wherein acoustic cue components of the first output signal match corresponding acoustic cue components of the first combination of audio signals.

The application discloses an audio signal adjustment method, device, equipment, medium and program product, and belongs to the field of audio processing. The method is executed by a client, and the method comprises the following steps: sending a reference audio signal to an airborne device in a target vehicle; recording a recorded audio signal, the recorded audio signal being a signal recorded by the client in a cabin of the target vehicle after the airborne device plays the reference audio signal; adjusting an audio filter with the same target of the recorded audio signal and the reference audio signal; and filtering a first audio signal played subsequently through the audio filter to obtain a second audio signal. The method records the recorded audio signal, the recorded audio signal being an audio signal associated with the reference audio signal, adjusts the audio filter with the same target of the two audio signals, and thus can filter and optimize the first audio signal played subsequently, so that the second audio signal is more suitable for listening in the current cabin.

A system for synchronizing and spatializing media, comprising client device and server device. A method for operation includes: tracking the real-time spatial coordinates of both devices via wireless sensors; synchronizing media playback using a timestamp-based protocol that compensates for network latency and clock offset; and applying directional audio filters, such as Head-Related Transfer Functions (HRTF), on the client device to render audio appearing to originate from server device's physical location. The method is further characterized by compensating for acoustic propagation and signal processing delays to ensure accurate spatiotemporal alignment. The system may be enhanced by a multimodal Al configured to perform operations such as real-time voice-preserving translation and the generation of synchronized, spatialized haptic feedback based on semantic event markers.

A computer-implemented method when executed by data processing hardware causes the data processing hardware to perform operations. The operations include receiving, from a sensor array, an audio signal at a unified post-filter, converting, via a conversion function, the audio signal into a Short-Time Fourier Transform (STFT) domain, determining, based on the converted audio signal, a speech-presence probability, and determining, based on the speech-presence probability, a noise smoothing factor. The operations also include estimating, via the unified post-filter, a noise power spectral density based on the noise smoothing factor, estimating, via the unified post-filter, a steering vector of a desired source, and generating, via the unified post-filter, a directionality-based mask and a coherence-based mask. The operations further include generating, based on the directionality-based mask and the coherence-based mask, a residual echo spectrum estimation and setting one or more spectral shaping factors based on the residual echo spectrum estimation.

Described is a method performed by a computation device for generating a binaural audio stream, comprising: receiving an audio stream for a sound source; determining a measure of processing capability of the computation device; selecting, based on the determined measure, a filtering mode from among a predefined set of filtering modes for use in an audio filtering process intended to convert the audio stream into a binaural audio stream; determining, based on a relative position of the virtual source location to a virtual listener location in a virtual listening environment, filter parameters for a set of filters specified by the selected filtering mode; generating the binaural audio stream by applying the audio filtering process to the audio stream, using the set of filters specified by the selected filtering mode; and outputting the binaural audio stream for playback. Further described are corresponding computation devices, computer programs, and computer-readable storage media.

A multiple-input, multiple-output audio processing is implemented as a linear system for use in an audio filter bank to convert a set of frequency domain input audio signals to a set of frequency domain output audio signals. A transfer function from one input to one output is defined as a frequency dependent gain function. In some implementations, the transfer function includes a direct component substantially defined as a frequency dependent gain and one or more decorrelated components having a frequency varying group phase response. The transfer function is formed by a set of sub-band functions, wherein each sub-band function is formed by a corresponding set of component transfer functions including a direct component and one or more decorrelated components.

Apparatus, systems, and methods for audio and video filtering for electronic user devices are disclosed. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute instructions to detect a visual event based on image data, the visual event representative of an activity associated with a likelihood of noise, the image data associated with a video stream output by a camera associated with a user device, and in response to the detection of the visual event, apply an audio filter to a portion of an audio stream corresponding to the image data in the video stream.

This invention provides a speech echo cancellation system, relating to the field of audio signal processing technology. The system includes acquiring target audio data and reference audio data and importing them into an audio storage buffer area; reading the target audio data based on an FIR echo cancellation filter and performing convolutional noise cancellation processing using the audio filtering coefficients of the FIR echo cancellation filter to generate audio echo cancellation data; if the audio echo cancellation data meets preset filtering conditions, calculating the cancellation error between the audio echo cancellation data and the reference audio data; if the cancellation error meets preset update conditions, using an adaptive algorithm to calculate the audio filtering coefficient increment based on the cancellation error and updating the audio filtering coefficients; and performing echo component cancellation processing on the acquired current audio data using the FIR echo cancellation filter, outputting the current echo cancellation data to the playback end, thereby enabling accurate identification and cancellation of echo components in the audio signal.

An apparatus comprises means configured to: obtain respective two or more audio signals from two or more microphones; determining a first sound source direction parameter and a first sound source energy parameter in one or more frequency bands of the two or more audio signals based on the processing of the two or more audio signals; determining a second sound source direction parameter and a second sound source energy parameter in one or more frequency bands of the two or more audio signals based on the processing of the two or more audio signals; obtaining a region defining a direction and / or range for the filter; and generating a filter to be applied to the two or more audio signals, where a filter gain / attenuation parameter is generated based on a relationship of the region to the first sound source direction parameter, the first sound source energy parameter, the second sound source direction parameter, and the second sound source energy parameter.

An apparatus including circuitry configured to: obtain two or more audio signals from respective two or more microphones; determine, in one or more frequency band of the two or more audio signals, a first sound source direction parameter and first sound source energy parameter based on processing of the two or more audio signals; determine, in the one or more frequency band of the two or more audio signals, a second sound source direction parameter and second sound source energy parameter based on processing of the two or more audio signals; obtain a region defining a direction and / or range for a filter; and generate the filter to be applied to the two or more audio signals, wherein filter gain / attenuation parameters are generated based on the region in relation to the first sound source direction parameter, the first sound source energy parameter, the second sound source direction parameter and the second sound source energy parameter.

The invention discloses an audio somatosensory thick vibrator which comprises a shell, a vibrator core is fixedly arranged in the shell and at least used for receiving audio and generating vibration, a controller is fixedly installed on the outer wall of one side of the shell and at least used for controlling vibration of the vibrator core, a plurality of installation frames are fixed to the outer wall of the shell, and the installation frames are arranged on the outer wall of the shell. An audio filtering module is integrated in a controller of the vibrator core, and the audio somatosensory vibrator is suitable for one or more scenes of automobile entertainment cabins, office chairs, sofas, physiotherapy mattresses and the like. The vibrator can realize multi-scene application, such as automobile entertainment cabins, office chairs, sofas, physiotherapy mattresses and the like, provides 4D somatosensory cinema experience, somatosensory music massage and sound wave vibration physiotherapy functions, and improves user experience.

This disclosure presents a template-based intelligent video automatic synthesis method, system, device, and medium. The template-based intelligent video automatic synthesis method includes: automatically generating several segment information based on the original video and user-defined cover, title, main video, and back cover templates. These segment information includes cover, title, main video, and back cover information, and each segment is independently configured. The method also involves parallel processing of audio sources from these segment information using a constructed audio mixing pipeline, and obtaining mixed audio based on the audio filter corresponding to each segment and the processed audio sources. Finally, the method calls a target engine to process each segment of the segment information and the mixed audio, and automatically generates a synthesized video based on the resolution scaling strategy of each segment. This method enables efficient generation of structured videos in vertical domains, is simple to reuse, and has low cost.

A multiple-input, multiple-output audio processing is implemented as a linear system for use in an audio filter bank to convert a set of frequency domain input audio signals to a set of frequency domain output audio signals. A transfer function from one input to one output is defined as a frequency dependent gain function. In some implementations, the transfer function includes a direct component substantially defined as a frequency dependent gain and one or more decorrelated components having a frequency varying group phase response. The transfer function is formed by a set of sub-band functions, wherein each sub-band function is formed by a corresponding set of component transfer functions including a direct component and one or more decorrelated components.

To provide a method for converting a set of frequency-domain input audio signals to a set of frequency-domain output audio signals for a sound system.SOLUTION: In a linear mixing system 100, a multi-input, multi-output audio process is implemented as a linear system for use in an audio filterbank to convert a set of frequency-domain input audio signals into a set of frequency-domain output audio signals. A transfer function from one input to one output is defined as a frequency dependent gain function. In some implementations, the transfer function includes a direct component that is substantially defined as a frequency dependent gain and one or more decorrelated components that have frequency-varying group phase response and is formed from a set of sub-band functions, with each sub-band function being formed from a set of corresponding component transfer functions including the direct component and one or more decorrelated components.SELECTED DRAWING: Figure 1

A computer-implemented method executed by data processing hardware causes the data processing hardware to perform operations that include receiving multiple audio signals from a sensor array. The multiple audio signals include a target audio signal and interference audio signals. The data processing hardware then identifies a design constraint based on the multiple audio signals. The desired constraint includes a pass constraint corresponding to the target audio signal and a null constraint corresponding to the interference audio signals. The data processing hardware then compares a design filter weight of the design constraint with a filter weight maximum, designs an audio filter using the desired constraint, and filters the multiple audio signals using the designed audio filter.

An apparatus includes components configured to perform the following operations: acquiring two or more corresponding audio signals from two or more microphones; determining a first sound source direction parameter and a first sound source energy parameter in one or more frequency bands of the two or more audio signals based on processing of the two or more audio signals; determining a second sound source direction parameter and a second sound source energy parameter in one or more frequency bands of the two or more audio signals based on processing of the two or more audio signals; obtaining a region defining the direction and / or range for a filter; and generating a filter to be applied to the two or more audio signals, wherein filter gain / attenuation parameters are generated based on the relationship between the region and the first sound source direction parameter, the first sound source energy parameter, the second sound source direction parameter, and the second sound source energy parameter.

A method of generating a substitution frame for a lost audio frame of an audio signal is presented. The method may comprise determining an audio filter based on samples of a valid audio frame preceding the lost audio frame. The method may comprise generating the substitution frame based on the audio filter and the samples of the valid audio frame preceding the lost audio frame. The method may be advantageously applied to a low frequency effects (LFE) channel of a multi-channel audio signal.