70results about "Tone control" patented technology

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.

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.

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.

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 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 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.

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.

A method for improving sound quality of a speaker includes: performing frequency division on an input signal of the speaker to obtain a first low-frequency input signal and a first high-frequency input signal, where the input signal is a time domain signal; performing transient detection on the first low-frequency input signal to determine whether the first low-frequency input signal is a transient signal; if the first low-frequency input signal is a transient signal, performing signal envelope modulation on the first low-frequency input signal to obtain a second low-frequency input signal; determining an output signal of the speaker based on the second low-frequency input signal and the first high-frequency input signal, where a startup voltage of the second low-frequency input signal is higher than a startup voltage of the first low-frequency input signal, and loudness of the second low-frequency input signal is greater than loudness of the first low-frequency input signal.

An electronic device includes a conductive lead, a semiconductor die, a package structure enclosing the semiconductor die and a portion of the conductive lead, and a non-conductive die attach film extending between the conductive lead and the semiconductor die and having a thickness less than 50 μm. A method of fabricating an electronic device includes singulating portions of a non-conductive die attach film on a carrier, partially singulating prospective die areas from a front side of a wafer, removing wafer material from a back side of the wafer to separate a semiconductor die from the wafer, and attaching a backside the semiconductor die to a singulated portion of the non-conductive die attach film on the carrier.

An electronic device (100) includes a conductive lead (107), a semiconductor die (110), a package structure (108) enclosing the semiconductor die (110) and a portion of the conductive lead (107), and a non-conductive die attach film (109) extending between the conductive lead (107) and the semiconductor die (110) and having a thickness (T1, T2) less than 50 μm. A method of fabricating an electronic device (100) includes singulating portions of a non-conductive die attach film (109) on a carrier, partially singulating prospective die areas from a front side of a wafer, removing wafer material from a back side of the wafer to separate a semiconductor die from the wafer, and attaching a backside the semiconductor die (110) to a singulated portion of the non-conductive die attach film (109) on the carrier.

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

The invention relates to an audio effect selection device (210) for processing an audio signal, comprising: - a memory (212) arranged to store datasets, each associated with a respective audio ambience, each dataset 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 ambience, and to select, as audio effects to be applied to an audio signal from said audio file, the audio effects from the dataset associated with the audio ambience for which the distance is the smallest. [Fig. 2]

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.

A network transceiver device is provided, including at least two variable gain amplifiers (VGAs), and at least two sets of analog digital converters (ADCs), each set including ADCs coupled to an output of one of the VGAs, the sets being arranged in VGA-specific channels. The device includes a plurality of feed-forward equalizers (FFEs), each FFE being coupled to receive an output of one of the ADCs in one of the VGA-specific channels. Each FFE is configured to adaptively equalize the output received from the ADCs utilizing a first equalization coefficient subset with coefficient values that are common to all FFEs, and a second equalization coefficient subset that is channel specific and that has a first set of coefficient values for a first VGA-specific channel and a second set of coefficient values for a second VGA-specific channel, the sets of coefficient values being computed independently.

An example apparatus (108A) includes: variable gain amplifier (VGA) circuitry (204) having: an input terminal, an output terminal, and a control terminal; and peak detector circuitry (222) having: an input terminal coupled to the output terminal of the VGA circuitry; an output terminal coupled to the control terminal of the VGA circuitry; the peak detector circuitry configured to: compare a voltage on the output (VOUT(P) - VOUT(N)) of the VGA circuitry to an offset voltage; and adjust, responsive to the comparison, a gain of the VGA circuitry.

An example apparatus (108A) includes: variable gain amplifier (VGA) circuitry (204) having: an input terminal, an output terminal, and a control terminal; and peak detector circuitry (222) having: an input terminal coupled to the output terminal of the VGA circuitry; an output terminal coupled to the control terminal of the VGA circuitry; the peak detector circuitry configured to: compare a voltage on the output (VOUT(P) - VOUT(N)) of the VGA circuitry to an offset voltage; and adjust, responsive to the comparison, a gain of the VGA circuitry.

An aspect includes a filtering method including operating a first filter to filter a first input signal to generate a first output signal; operating a second filter to filter a second input signal to generate a second output signal; and selectively coupling at least a portion of the second filter with the first filter to filter a third input signal to generate a third output signal. Another aspect includes a filtering method including operating switching devices to configure a filter with a first set of pole(s); filtering a first input signal to generate a first output signal with the filter configured with the first set of pole(s); operating the switching devices to configure the filter with a second set of poles; and filtering a second input signal to generate a second output signal with the filter configured with the second set of poles.

To provide a display system having an audio output device.SOLUTION: The display system includes a head-mounted display unit and a detachable speaker unit. The head-mounted display unit outputs visual content to the user and provides the user with a visual pass-through of the real environment. The detachable speaker unit may be detachably coupled to the head-mounted display unit to provide auditory content to the user. At least one of the visual content or the audio content is modified according to a position of the detachable speaker unit relative to the head-mounted display unit.SELECTED DRAWING: Figure 1A

An electronic device outputting a sound according to embodiments of the disclosure may comprise memory storing at least one instruction, a processor executing the at least one instruction, at least one speaker, and an audio processing module processing the input audio signal and outputting the processed input audio signal to the at least one speaker. The audio processing module may include a digital amplifier amplifying a digital audio signal, a digital-to-analog converter (DAC) converting the digital audio signal into an analog audio signal, an analog amplifier amplifying the analog audio signal, a boost amplifier boosting the analog audio signal to a reference high voltage or more, and at least one noise gate removing noise included in the analog audio signal. The digital amplifier may apply a positive (+) digital gain to the digital audio signal. The analog amplifier may apply a negative (−) analog gain to the analog audio signal.

The invention provides an adaptive equalizer circuit and an operation method thereof. The adaptive equalizer circuit includes an amplifier circuit and an adaptive circuit. The amplifier circuit is coupled to the adaptive circuit. The amplifier circuit receives an input signal. The amplifier circuit provides a compensation signal to the adaptive circuit based on an input signal. The adaptability circuit provides a compensation setpoint and a feedback signal to the amplifier circuit based on the compensation signal. A low-frequency response and a high-frequency gain of the amplifier circuit are adjusted based on the feedback signal and the compensation setpoint, respectively.

Systems are provided for controlling audio content to a user by a personal audio device. A method includes: receiving sensor data from a sensor of the personal audio device; determining, by a processor, a mental state of the user based on the sensor data; determining, by the processor, at least one audio setting based on the mental state; and controlling, by the processor, audio signals of the personal audio device based on the at least one audio setting.

To provide a method, a computing system and a non-temporary computer readable storage medium for adjusting audio playback setting based on analysis of audio characteristics.SOLUTION: A media unit 106 of a vehicle includes: an EQ model query generator for generating an equalization (EQ) query including expression of a sample of an audio signal to a neural network; an EQ filter setting analyzer for accessing a plurality of audio playback settings determined by the neural network based on the EQ query, and determining an EQ filter coefficient to apply to the audio signal based on the plurality of audio playback settings; and an EQ adjusting implementer for applying the filter coefficient to the audio signal during a first duration.SELECTED DRAWING: Figure 2

A circuit (100) includes a programmable gain amplifier (PGA 102) having a PGA output. The circuit (100) further includes a delta-sigma modulator (104) having an input coupled to the PGA output. The circuit (100) also includes a digital filter (114) and a dynamic range enhancer (DRE) circuit (110). The digital filter (114) is coupled to the delta-sigma modulator output and to the PGA (102). The DRE circuit (110) is configured to monitor a signal level of the delta-sigma modulator output. In response to the signal level being less than a DRE threshold, the DRE circuit (110) is configured to program the PGA to a gain level greater than unity gain and to cause the digital filter to implement an attenuation of the same magnitude as the gain level to be programmed into the PGA (102).

A device comprises a discrete-time linear equalizer circuit. The discrete-time linear equalizer circuit includes a sample-and-hold switching logic that incorporates several switched-capacitor circuits. The multiple switched-capacitor circuits include at least one switched-capacitor circuit for a precursor tap, one switched-capacitor circuit for a cursor tap, and one switched-capacitor circuit for a trailing tap.A clock-controlled switching logic circuit is used to connect a capacitor in the circuit with the switched capacitor of the predecessor tap to a negative signal input for a first period, a capacitor in the circuit with the switched capacitor of the cursor tap to a positive signal input for a second period, and a capacitor in the circuit with the switched capacitor of the trailing tap to the negative signal input for a third period. The clock-controlled switching logic circuit is also used to connect the capacitors of the circuits with the switched capacitors in parallel for a fourth period.

An audio-signal equalisation device for a vehicle using a data communication bus includes a memory storing speed filter data sets comprising triplets associating a speed value and a gain value associated with coefficients for defining a speed masking filter, and functional filter data sets comprising triplets associating an engine or ventilation value and a gain value associated with coefficients for defining an engine or ventilation masking filter. The device further includes a collector receiving parameters accessible on the communication bus and designating the vehicle speed and a functional parameter selected from the vehicle engine speed and a ventilation data set. The device further includes a computer receiving as an input a noise margin value, and the parameters, to derive therefrom a speed filter data set and a functional filter data set and to combine them in a masking equalisation filter to be applied to the audio signal.

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 present application provides a method and device for improving the sound quality of a loudspeaker, relates to the technical field of media, and can improve the low-frequency sound effect of the loudspeaker and improve the sound quality of the loudspeaker. The method comprises the following steps: performing frequency division on an input signal of the loudspeaker to obtain a first low-frequency input signal and a first high-frequency input signal, wherein the input signal is a time-domain signal; performing transient detection on the first low-frequency input signal to determine whether the first low-frequency input signal is a transient signal; if the first low-frequency input signal is a transient signal, performing signal envelope modulation on the first low-frequency input signal to obtain a second low-frequency input signal; and then determining an output signal of the loudspeaker according to the second low-frequency input signal and the first high-frequency input signal. The starting voltage of the second low-frequency input signal is greater than the starting voltage of the first low-frequency input signal, and the loudness of the second low-frequency input signal is greater than the loudness of the first low-frequency input signal.

An aspect of the present disclosure is directed to systems, methods, and software products that enhance the sound and perception of bass energy in loudspeaker systems which have limited response in the low frequency spectrum. The systems, methods, and software products provide for (i) receiving one or more bass sub-bands of audio channel content, wherein each bass sub-band is separated from an audio program content spectrum including high-band energy; (ii) applying boost to each bass sub-band to compensate for rolloff of the speaker system; (iii) applying a limiter to prevent speaker saturation at low frequencies and / or high volume levels; and (iv) combining each bass sub-band along with the high band energy for sending as a recombined audio signal to the speaker system for rendition as physical sound.

Provided are an information processing device and an information processing method capable of correcting the frequency characteristics of sound output from a speaker and making it easier to hear the sound clearly. The information processing apparatus includes: a volume level information acquisition unit that acquires volume level information relating to a setting of a volume level by a user; and a signal processing unit that corrects the frequency characteristic of the sound output from the speaker, on the basis of the volume level, such that the attenuation amount from the maximum volume of the sound output from the speaker at a prescribed frequency falls within a volume lower limit value range with respect to the maximum volume at the prescribed frequency, and corrects the frequency characteristic of the sound output from the speaker, on the basis of the volume level. The volume lower limit value is based on the difference between the maximum volume of the speaker at the reference frequency and the minimum volume that can be set by the sound output from the speaker. The sound pressure level difference is determined according to a first sound pressure level difference between the maximum volume under the reference frequency and the minimum audible volume which can be heard by people and a second sound pressure level difference between the maximum volume and the minimum audible volume under other frequencies different from the reference frequency.

A system and apparatus for generating vibrations of a vehicle seat synchronized with music output from an audio device, the system and apparatus using an electric motor included in the seat to adjust the seat in forward and backward directions and in upward and downward directions, the system comprising: an electric motor multi-control circuit configured to receive signals from a seat position adjustment switch and an audio switch, generate a drive current required to vibrate the seat, and output the drive current to the electric motor. The electric motor receives signals from the electric motor multi-control circuit and generates motor torque while being driven, and the generated motor torque is transmitted to the seat to generate the target vibration in the seat.