162results about "Tone control" patented technology

In various embodiments, a sound modification application selectively modifies one or more sounds included in one or more audio signals. In operation, the sound modification application determines classifications associated with multiple sounds included in one or more audio signals. The sound modification application selects a first frequency sub-band of a first sound included in the multiple sounds based on a first classification associated with the first sound. The sound modification application then modifies the first frequency sub-band of the first sound, without modifying at least a second frequency sub-band of the first sound, to generate a modified audio signal.

According to one embodiment, a low noise amplifier (LNA) circuit includes a first stage which includes: a first transistor; a second transistor coupled to the first transistor; a first inductor coupled in between an input port and a gate of the first transistor; and a second inductor coupled to a source of the first transistor, where the first inductor and the second inductor resonates with a gate capacitance of the first transistor for a dual-resonance. The LNA circuit includes a second stage including a third transistor; a fourth transistor coupled between the third transistor and an output port; and a passive network coupled to a gate of the third transistor. The LNA circuit includes a capacitor coupled in between the first and the second stages, where the capacitor transforms an impedance of the passive network to an optimal load for the first amplifier stage.

An audio device is provided. The audio device comprises a controller and an acoustic transducer. The controller is configured to receive an input audio signal. The controller is further configured to generate, via a loudness detector, a loudness level of a first portion of the input audio signal. The controller is further configured to identify a compression curve of a plurality of compression curves corresponding to a user volume setting. The controller is further configured to adjust, via an audio compressor, a second portion of the input audio signal based on the loudness level and the compression curve to generate an output audio signal. The loudness level is determined according to LUFS model.

The embodiment of the invention provides an impedance device and a system for simulating the vibration influence of a head on a vibration unit, the impedance device can comprise a mass part, an elastic part and a fixing part, the mass part is connected to the fixing part through the elastic part, the fixing part is a structure body with a hollow interior, the fixing structure comprises an opening, and the opening is communicated with the elastic part. The mass part is provided with an opening, the elastic part is located at the opening and connected with the fixed part, the elastic part and the fixed part form a cavity, and the elastic coefficient of the elastic part in the vibration direction of the mass part relative to the fixed part ranges from 600N / m to 5000N / m.

An analog front end (AFE) circuit including: a continuous time linear equalizer (CTLE) circuit; a transimpedance amplifier (TIA) connected to the CTLE circuit; and a feedback circuit including: a first transistor connected between a first output of the feedback circuit and a first node connected to a first current source; a second transistor connected between a second output of the feedback circuit and a second node connected to a second current source; and a first tunable resistor coupled between the first node and the second node, wherein: a first input of the feedback circuit is connected to a first output of the TIA; a second input of the feedback circuit is connected to a second output of the TIA; the second output of the feedback circuit is connected to a first input of the TIA.

A method and system of neural network dynamic noise suppression is provided for audio processing.

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.

Generative media content (e.g., generative audio) can be played back across multiple playback devices concurrently. A generative content group coordinator device can receive input parameters, which can include sensor data, media content, or other such input. The coordinator device can generate first and second generative media content streams, each of which can be transmitted to first and second playback devices, respectively. The first and second playback devices can play back the first and second streams of generative media content concurrently.

The invention relates to a device (210) for selecting audio effects for processing an audio signal comprising: - a memory (212) arranged to store data sets each associated with a respective audio ambiance, each data set comprising a vector of reference values each corresponding to a respective audio component, a vector of weighting coefficients each corresponding to a respective audio component, and audio effects, and - a calculator (214) arranged to receive a vector of values associated with an audio file, each value corresponding to a respective audio component, to calculate a respective distance between said audio file and each audio ambiance and to select, as audio effects to be applied to an audio signal taken from said audio file, the audio effects of the data set associated with the audio ambiance for which the distance is the smallest. [Fig. 2]

To provide methods and systems for performing automatic speed-based audio control.SOLUTION: One method includes: receiving, with an electronic control unit included in a vehicle, a speed of the vehicle; and receiving, with the electronic control unit, an audio signal. The method also includes: accessing, with the electronic control unit, multiple equalization curves based on the speed of the vehicle, each of the multiple equalization curves being associated with the speed of the vehicle, and each of the multiple equalization curves defining a gain adjustment for one of multiple frequencies; and, for each curve of the multiple equalization curves, applying the gain adjustment defined by the curve to one of the multiple frequencies of the audio signal.SELECTED DRAWING: Figure 2

A computer-implemented method, apparatus, and computer program product for a sound system. Speech recognition is performed on input audio data including a speech input to the sound system. Additionally, speech recognition is performed on at least one instance of output audio data including speech reproduced by one or more audio speakers of the sound system. A difference between a result of the speech recognition performed on the input audio data and a result of the speech recognition performed on the corresponding instance of the output audio data is determined. When the difference is greater than or equal to a threshold, the quality of the reproduced speech is determined to be unacceptable. If it is determined that the speech quality of the reproduced sound is unacceptable, a corrective action may be performed to improve the quality of the speech reproduced by the sound system.

A display system includes a head-mounted display unit and a detachable speaker unit. The head-mounted display unit outputs visual content to a user and provides a visual pass-through of a real environment to the user. The detachable speaker unit is detachably coupleable to the head-mounted display unit for providing aural content to the user. At least one of the visual content or the aural content is changed according to a position of the detachable speaker unit relative to the head-mounted display unit.

Generative media content (e.g., generative audio) can be played back across multiple playback devices concurrently. A generative content group coordinator device can receive input parameters, which can include sensor data, media content, or other such input. The coordinator device can generate first and second generative media content streams, each of which can be transmitted to first and second playback devices, respectively. The first and second playback devices can play back the first and second streams of generative media content concurrently.

A display system with an audio output device is disclosed. For example, a display system including a head-mounted display unit and a removable speaker unit is disclosed. The head-mounted display unit outputs visual content to the user and provides visual delivery of the real environment to the user. The detachable speaker unit can be detachably coupled to the head-mounted display unit for providing auditory content to a user. At least one of the visual content or the auditory content changes according to a position of the detachable speaker unit relative to the head-mounted display unit.

Techniques for dynamically and automatically equalizing audio content are provided. These techniques include receiving, in portions or as a whole, an instruction file that contains a plurality of instruction lines. Each one of the instruction lines correlates an equalizer setting instruction with a time frame of media content. The techniques further include automatically adjusting, as the media content is played, an equalizer setting of the equalizer according to the equalizer setting instruction that corresponds to the time frame of the media content being played at that time. Audio content is equalized in accordance with the equalizer setting, which is automatically adjusted as the media content is played according to the instruction file.

The present invention provides an envelope tracking power modulator, which includes an amplifier circuit and a zero peak circuit. The amplifier circuit receives an envelope input, generates a modulated power supply voltage according to the envelope input, and provides the modulated power supply voltage to a power amplifier. The zero peak circuit is coupled to the amplifier circuit and applies a zero peak value to the amplifier circuit, wherein the zero peak value inserts a zero point at a frequency. Accordingly, the present invention also provides a wireless communication system and an envelope tracking calibration method, wherein the zero peak circuit inserts a zero point at a frequency to increase the bandwidth of envelope tracking without requiring additional static current consumption.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

To provide a microphone capable of setting a correction value without need for a sound source.SOLUTION: A microphone includes a MEMS device 200, a driver 330 and a control section 320. The MEMS device outputs a first electric signal corresponding to an acoustic pressure. The driver vibrates the MEMS device through a driving signal. The control section computes a correction value for correcting the first electric signal on the basis of a second electric signal output from the MEMS device when being vibrated by the driving signal.SELECTED DRAWING: Figure 6

To enable a patron of a venue attending a performance to feel as if he or she is ‘front and center’ even if located much farther away or angled from a stage of the venue, system and methods that enhance audio quality of audio signals especially over the voice frequency spectrum to generate a processed audio signal including an increased voice frequency spectrum are provided. The processed audio signal may be communicated to a receiving system with a headset or other transducer (e.g., loudspeakers) that allows for ambient sounds to be heard by a patron and for the patron to control volume of the processed audio signal, thereby mixing the processed audio signal with the ambient sounds. The receiving system may be rented and an operator may control usage for both performance or long-term rentals using various control features of the processed audio signals and / or receiving system.

The various implementations described herein include methods, devices, and systems for automatic audio equalization, In one aspect, a method is performed at a computing system that includes speaker(s), microphones, processors and memory. The computing system outputs audio user content and automatically equalizes the audio output of the computing system. The equalizing includes: (1) receiving the outputted audio content at each microphone of the plurality of microphones; (2) based on the received audio content, determining an acoustic transfer function for the room; (3) based on. the determined acoustic transfer function, obtaining a frequency response for the room; and (4) adjusting one or more properties of the speakers based on the determined frequency response.

The audio device includes a receiver 101 configured to receive an audio signal from which an audio beamformer 103 generates a beamformed output audio signal. The beamformer 103 includes a signal processor 201 that applies signal processing to the audio signal. Signal processing includes spatial beamforming and spatial decorrelation, which is often adapted spatial decorrelation. A decorrelation adapter 203 adapts decorrelation parameters depending on the audio signal, and a beamform adapter 205 adapts beamform parameters depending on the beamformed output audio signal. An equalizer 105 applies frequency equalization to the beamformed output audio signal to generate an output signal, and an equalization adapter 107 adapts frequency equalization depending on the beamform parameters, typically also depending on the decorrelation parameters. This approach provides improved fidelity to the desired sound source and attenuates interfering sound sources.

The invention relates to a method (100) for processing audio signals, comprising the following method steps - Providing (110) an audio signal x(n); - Providing (120) a filter function; - applying (130) the filter function to the audio signal x(n) in the time domain to provide a filtered audio signal x̃(n); and - outputting (140) the filtered audio signal x̃(n); where - the provision of the filter function is dependent on K individual filter functions, where K ≥ 2; and - wherein the individual filter functions comprise at least two high-pass filter functions or at least two low-pass filter functions from which a band-pass filter function is determined.

In various embodiments, a sound modification application selectively modifies one or more sounds included in one or more audio signals. In operation, the sound modification application determines a classification associated with a plurality of sounds included in the one or more audio signals. The sound modification application selects a first frequency subband of a first sound included in the plurality of sounds based on a first classification associated with the first sound. The sound modification application then modifies the first frequency subband of the first sound while not modifying at least a second frequency subband of the first sound to generate a modified audio signal.

The present disclosure provides systems and methods for progressively transitioning a playback volume based on a background noise level. An apparatus (800) can have a predetermined amount of volume levels between a minimum volume and a maximum volume. The volume levels can be spaced apart by a predetermined increment such that there is a predefined difference between adjacent volume levels. A digital gain (102) can be used to adjust the volume levels, and an analog gain (106) can allow the playback volume to be adjusted to an amount between adjacent volume levels. The playback volume can progressively transition within a predefined volume level by an amount less than the predetermined difference between adjacent volume levels. This smaller amount can be the analog gain (106).

The application discloses a method for optimizing the working state of a bone conduction earphone, characterized in that the bone conduction earphone comprises an earphone core and at least one vibration sensor, and the method comprises the following steps: acquiring a vibration signal through the at least one vibration sensor, wherein the vibration signal is at least partially derived from the vibration generated by the earphone core in response to an audio signal, and the vibration of the earphone core is configured to be transmitted to a user wearing the bone conduction earphone in a bone conduction manner; determining the vibration response characteristics of the earphone core based on the vibration signal and the audio signal; and feeding back the working state of the bone conduction earphone based on the vibration response characteristics of the earphone core.

Disclosed are processes, methods, systems, and devices for intelligently detecting speech or audio signals and smoothly transitioning to a mode that allows a user to better understand the speech. For example, aspects of the present disclosure provide a method for processing and generating an audio signal. During playback of the audio signal, the method analyzes the content of the audio signal before playback of the content to determine whether one or more predefined conditions are met to indicate that the content includes speech. In response to determining that the one or more predefined conditions are met, the method automatically applies a first playback equalization to the audio signal, the first playback equalization being configured to enhance the speech in the content. In response to determining that the one or more predefined conditions are not met, the method includes applying no playback equalization to the audio signal or applying a second playback equalization different from the first playback equalization.

The present disclosure relates to a voltage gain amplifier for automotive radar. Disclosed herein is a voltage gain amplifier for use in an automotive radar receiver chain. The voltage gain amplifier utilizes pole-to-zero cancellation to yield a desired transfer function without gain peaking at a bandwidth where peaking is desired, and utilizes a low pass filter that is effectively formed by a feedback loop that includes a high pass filter and a differential amplifier to ensure a desired level of attenuation at the desired bandwidth. In some instances, a chopper can be utilized in the feedback loop before the high pass filter and after the differential amplifier in order to reduce the bandwidth of the differential amplifier in the feedback loop.

An audio signal processing device includes: an obtainer that obtains an audio signal; a first signal processor that generates, based on the audio signal, a first high-frequency signal including a high-frequency component in a frequency band higher than a first frequency higher than a human hearing range; and an adder that adds the first high-frequency signal to the audio signal to generate an expanded audio signal having an expanded high-frequency range. The first signal processor generates the first high-frequency signal by performing: (i) first signal processing including first clipping processing of clipping the audio signal at a first amplitude; and (ii) first high-pass filter processing of passing an audio signal obtained by the first clipping processing through the frequency band higher than the first frequency.

To stabilize a location of a sound image.SOLUTION: In a side of quadrangle on a spherical surface in which four speakers surrounding a target sound image position are set as a peak, it is assumed that the virtual speaker is positioned onto the side positioned at a lower side. By performing a three-dimension VBAP from the virtual speaker and the two speakers on an upper right and an upper left, each gain of the two speaker on the upper right and the upper left for positioning a sound image to the target sound image position and the virtual speaker is calculated. Further, by performing a secondary VBAP of the speakers in a lower right and a lower left, each gain of the speakers on the lower right and the lower left for positioning the sound image to the virtual speaker is calculated. The gain obtained by multiplying the gain of the virtual speaker to the gain is a gain of the speaker in the lower right and the lower left for positioning the sound image to the target sound image position. The present technique is adapted to the voice processing device.SELECTED DRAWING: Figure 7

According to one embodiment, a low noise amplifier (LNA) circuit includes a first stage which includes: a first transistor; a second transistor coupled to the first transistor; a first inductor coupled in between an input port and a gate of the first transistor; and a second inductor coupled to a source of the first transistor, where the first inductor and the second inductor resonates with a gate capacitance of the first transistor for a dual-resonance. The LNA circuit includes a second stage including a third transistor; a fourth transistor coupled between the third transistor and an output port; and a passive network coupled to a gate of the third transistor. The LNA circuit includes a capacitor coupled in between the first and the second stages, where the capacitor transforms an impedance of the passive network to an optimal load for the first amplifier stage.