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Небесная энциклопедия

Космические корабли и станции, автоматические КА и методы их проектирования, бортовые комплексы управления, системы и средства жизнеобеспечения, особенности технологии производства ракетно-космических систем

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Мониторинг СМИ

Мониторинг СМИ и социальных сетей. Сканирование интернета, новостных сайтов, специализированных контентных площадок на базе мессенджеров. Гибкие настройки фильтров и первоначальных источников.

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Поддерживает ввод нескольких поисковых фраз (по одной на строку). При поиске обеспечивает поддержку морфологии русского и английского языка
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Применить Всего найдено 1964. Отображено 100.
02-02-2012 дата публикации

Bell sound outputting apparatus and method thereof

Номер: US20120027225A1
Автор: Jong-youb RYU
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A method and apparatus for outputting a bell sound are disclosed. The method includes receiving an ambient noise; extracting a target frequency band from the ambient noise; changing volume of a pre-stored bell sound by using the target frequency band; and outputting the changed pre-stored bell sound. In the apparatus for outputting a bell sound, when a bell sound outputting apparatus receives an incoming call signal, a user may hear a bell sound including a signal in a frequency band not existing in a peripheral environment, and thus may easily recognize the bell sound.

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Номер записи: 1
22-03-2012 дата публикации

method and apparatus for providing information about the source of a sound via an audio device

Номер: US20120071997A1
Автор: Dzmitry Viktorovich Aliakseyeu, Igor Berezhnoy
Принадлежит: KONINKLIJKE PHILIPS ELECTRONICS NV

A method and apparatus for providing information about the source of a sound via an audio device is described. An ambient sound is detected ( 200 ) and specific sounds are identified in the detected ambient sound ( 202 ). Information about the source of the identified specific sounds is determined ( 204 ). An operational control characteristic of a generated audio stream rendered by an audio device is changed ( 206 ) and information about the source is provided to the audio device upon detection of said identified specific sounds ( 208 ).

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Номер записи: 2
28-06-2012 дата публикации

Original equipment manufacturer ("oem") integration amplifier

Номер: US20120163621A1
Автор: Brett D. Riggs, Nathaniel Lee Wincek, Sam W. Grove
Принадлежит: AAMP of America

Embodiments of the proposed invention preferably provide systems and methods for an improved aftermarket automobile audio amplifier that is easy to install while integrating and improving OEM functionality. Embodiments of the invention preferably directly connect to preexisting OEM wiring and hardware. Embodiments of the invention preferably preserve OEM system functionality and allow for integration of additional auxiliary inputs. A digital signal processing module preferably adjusts an output volume to ensure that auxiliary inputs are audible to a vehicle occupant at an appropriate and comfortable volume level. Therefore, vehicle OEM safety chimes or third party navigational commands may be output to a vehicle occupant at an appropriate volume level irrespective of the presence, absence or volume level of other audio signal outputs. Furthermore, the direct connection to preexisting OEM wiring and hardware preferably allows for both easy installation and easy removal of embodiments of the current invention.

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Номер записи: 3
14-03-2013 дата публикации

Information processing apparatus and information processing method

Номер: US20130064396A1
Автор: Naoki Sugita, Takashi Tsurumoto
Принадлежит: Sony Corp

An information processing apparatus including an audio input unit, an amplifier unit, and a controller. The audio input unit is capable of converting audio into an electric audio signal. The amplifier unit is configured to amplify the audio signal obtained by the audio input unit and capable of performing auto gain control. The controller is configured to turn off the auto gain control of the amplifier unit when an application that uses an audio recognition is executed with respect to the amplified audio signal.

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Номер записи: 4
15-08-2013 дата публикации

Method and apparatus to automatically control audio volume

Номер: US20130208919A1
Автор: Chi-ho CHUNG
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A method includes comparing a magnitude value of an audio signal in a fixed unit section with a target volume level, measuring audio gain in each fixed unit section using specific functions which are each determined to be different according to a ratio of the target volume level to the audio signal magnitude, and providing the measured audio gain to the audio signal in the fixed unit section.

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Номер записи: 5
31-10-2013 дата публикации

DIGITAL AUTOMATIC GAIN CONTROL

Номер: US20130285747A1
Автор: Kindred Jon S., Sorensen Bryant, Wahl Jerry
Принадлежит:

Systems, devices, and methods are provided to inhibit apparent amplitude modulation in non-linear processing that causes distortion in a processed signal. One aspect of the invention includes a hearing aid. The hearing aid includes a microphone to receive an input signal, a speaker to reproduce the input signal, and a processor. The processor processes the input signal using a gain. The processor includes an inhibitor, which inhibits distortions, and an adjuster, which adjusts the gain. The inhibitor acts to smooth an envelope of the input signal to inhibit undesired modulation. The adjuster adjusts the gain if the envelope is either above or below a threshold. 1. A digital analog gain control , comprising:a detector to detect an envelope of an input signal using Hilbert filters;an adder to provide a difference between the envelope and a threshold; andan adjuster that adjust a gain if the difference is one of two conditions, wherein the two conditions includes being greater than zero and being less than zero.2. The digital analog gain control of claim 1 , further comprising a filter that removes low frequencies claim 1 , wherein the filter receives the input signal claim 1 , removes frequencies less than about 100 Hertz from the input signal claim 1 , and presents the input signal to the detector.3. The digital analog gain control of claim 2 , further comprising a digital delay element that receives the input signal and presents a delayed input signal.4. The digital analog gain control of claim 3 , further comprising a first Hilbert filter and a second Hilbert filter claim 3 , wherein the first Hilbert filter receives the delayed input signal and filters the delayed input signal to form the first filtered input signal claim 3 , and wherein the second Hilbert filter receives the input signal and filters the input signal to form the second filtered input signal.539.-. (canceled)40. The digital analog gain control of claim 4 , further comprising:a first multiplier and ...

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Номер записи: 6
27-02-2014 дата публикации

Amplifier Bias Control

Номер: US20140055202A1
Автор: Hayes Kevin M.
Принадлежит:

An improved method for maintaining optimal amplifier bias current utilizing a signal conditioning element which serves to symmetrically condition a sense voltage such that the sense voltage distortion is substantially determined by properties of the signal conditioning element rather than by properties of the amplifier amplification devices or the input perturbing signal. 1. A system for controlling the bias current of an electronic gain element , comprising:a. an electronic gain element;b. a resistor connected to said electronic gain element such that the voltage across said resistor represents the current through said electronic gain element;c. a reference signal which is proportional to the optimal bias current of said electronic gain element;d. a signal conditioning element that limits said resistor voltage to a predetermined arbitrary high voltage level and to a predetermined arbitrary low voltage level to create a conditioned signal; ande. A low frequency control loop element to operate on said conditioned signal in conjunction with said reference signal to modify the operating voltages of said electronic gain element such that the optimal bias current is maintained.2. The system of claim 1 , wherein the electronic gain element is intended to operate in class A claim 1 , class AB and class B regions.3. The system of claim 1 , wherein the electronic gain element is a vacuum tube or a MOS JFET claim 1 , or bipolar transistor.4. The system of claim 1 , wherein the electronic gain element is a vacuum tube.5. The system of claim 1 , wherein the signal conditioning element uses a common mechanism to affect the positive going limiting and negative going limiting.6. The system of claim 1 , wherein the signal conditioning element is a clamping amplifier.7. The system of claim 1 , wherein the signal conditioning element is an Analog Devices AD8036 clamping amplifier.8. The system of claim 1 , wherein the control loop element has a bandwidth less than 20 Hz.9. A method ...

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Номер записи: 7
06-01-2022 дата публикации

GENERATING HARMONICS IN AN AUDIO SYSTEM

Номер: US20220006436A1
Автор: RÖJDER Johannes
Принадлежит:

A method and corresponding system for generating harmonics from an input signal having an input frequency uses a harmonics generator having an associated gain controller generating adjustable gain control factors, controlling the amplitude of harmonic components. The method includes generating a series of harmonic components in a process including adjustment of relative balances between at least three harmonic components of different orders, the balances being independently adjustable by controlling at least two gain control factors. Each gain control factor is adjusted based on a frequency dependent measure of the input signal or signal derived therefrom, of the frequency dependent measure of the input signal and/or derived signal unique for the corresponding gain control factor. An output signal is created including the harmonic components according to the balances as controlled by the at least two gain control factors, wherein the balances between the harmonic components depend on the input frequency. 1212214. A method for generating harmonics from an audio input signal having an input frequency by using a harmonics generator () having an associated gain controller () for generating adjustable gain control factors for controlling the amplitude of harmonic components , wherein the method comprises: wherein each particular gain control factor of said at least two gain control factors is adjusted based on at least one frequency dependent measure of the input signal and/or a signal derived therefrom, one or more of said at least one frequency dependent measure of the input signal and/or a signal derived therefrom being unique for corresponding particular gain control factor, wherein an intermediate signal corresponds to a signal derived from the input signal,', 'creating (S2) an output signal comprising said harmonic components according to said balances as controlled by said at least two gain control factors, wherein the balances between said at least three harmonic ...

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Номер записи: 8
08-01-2015 дата публикации

Electronic circuit arrangement for receiving low frequency electromagnetic waves with an adjustable attenuator element

Номер: US20150011171A1
Автор: Akbari-Dilmaghani Rahim
Принадлежит: MAXIM INTEGRATED PRODUCTS, INC.

An electronic circuit arrangement for receiving low-frequency electromagnetic waves is proposed, having an inductor (L) acting as an antenna for generating a received signal, having a first receiver (), connected to the inductor (L), for decoding a first component of the received signal and having a second receiver (), connected to the inductor (L), for decoding a second component of the received signal, wherein at least the second receiver () is connected to the inductor (L) via an attenuator element () having adjustable attenuation, wherein at least one adjustment signal generation circuit () is provided for generating an adjustment signal corresponding to a voltage of the received signal which is fed to the attenuator element () for adjusting the attenuation. 1. (canceled)2. An electronic circuit comprising:an antenna that receives a wireless signal and outputs an encoded signal;a first decoder coupled to receive at least a first portion of the first encoded signal, the first decoder decodes a first component of the encoded signal;a second decoder coupled to receive at least a second portion of the first encoded signal, the second decoder decodes a second component of the encoded signal and switches the electronic circuit between a standby mode and an operating mode, the second decoder operating at a lower voltage level than the first decoder; andan attenuator element coupled between the antenna and the second decoder, the attenuator element determines if the at least second portion of the first encoded signal is above a voltage threshold and reduces a voltage level on the at least one second portion of the first encoded signal if the voltage threshold is exceeded.3. The electronic circuit according to wherein the attenuator element comprises a voltage divider having a plurality of voltage reduction steps that can reduce the at least second portion of the first encoded signal below the voltage threshold.4. The electronic circuit according to wherein the voltage ...

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Номер записи: 9
11-01-2018 дата публикации

Method of Controlling an Audio Amplifier Unit and Audio amplifier Unit

Номер: US20180013394A1
Автор: Grell Axel, Grimm Alexander
Принадлежит: Sennheiser electronic GmbH & Co. KG

A method for controlling an audio amplifier unit having an audio input, an amplifier, a volume regulation unit and an audio output. An audio signal is received via the audio input and is amplified by the amplifier, and an adjustment procedure modifies an operating curve of the volume regulation unit for the received audio signal. From a modified setting of the volume regulation unit, a modified amplification value is calculated according to a first volume operating curve. The amplifier's amplification is modified to match the modified amplification value, and a second volume operating curve is determined and assigned to the calculation of modified amplification values of the amplifier. The modified amplification value is substantially at the center of the second volume operating curve. Further, the volume regulation unit is adjusted mechanically or electrically to a predefined position. 1. A method for controlling an audio amplifier unit , wherein the audio amplifier unit comprises an audio input , an amplifier , a volume regulation unit , and an audio output , the method comprising:amplifying an audio signal received via the audio input, wherein a first amplification of the amplifier is selected according to a first volume operating curve, which is a currently valid operating curve, by utilizing the volume regulation unit;detecting a modified setting of the volume regulation unit made by a user;calculating a modified amplification value from the modified setting of the volume regulation unit according to the first volume operating curve, wherein the modified amplification value corresponds to a modified volume;modifying the amplification of the amplifier to be the modified amplification value;automatically performing an adjustment procedure, wherein a second volume operating curve is determined wherein the second volume operating curve becomes the currently valid operating curve, and wherein said modified amplification value is substantially in the center of the ...

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Номер записи: 10
21-01-2016 дата публикации

Volume Control Assembly

Номер: US20160020741A1
Автор: Bell Paul
Принадлежит:

A volume control assembly includes a housing that may be positioned proximate an external electronic device. A processor is coupled to the housing. A loud actuator is coupled to the housing and the processor. The loud actuator determines a maximum volume of an audio signal. A quiet actuator is coupled to the housing and the processor. The quiet actuator determines a minimum volume of the audio signal. An amplifier is coupled to the housing and the processor. The amplifier amplifies the audio signal above the minimum volume and below the maximum volume. An input is coupled to the housing, the processor and the external electronic device. The input receives the audio signal from the external electronic device. An output is coupled to the housing, the processor and the external electronic device. The audio signal is directed to the external electronic device between the minimum and maximum volume.

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Номер записи: 11
17-01-2019 дата публикации

SYSTEM AND METHOD FOR DIGITAL SIGNAL PROCESSING

Номер: US20190020950A1
Автор: Bongiovi Anthony, Fuller Phillip, Zelniker Glenn
Принадлежит:

The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal. 1. A system for digital signal processing of an input audio signal comprising:a high pass filter configured to filter the input audio signal to create a high pass signal,a first filter module configured to create a first filtered signal,a first compressor configured to modulate the first filtered signal to create a modulated signal,a second filter module configured to create a second filtered signal,a first processing module configured to process the second filtered signal to create a processed signal,a band splitter configured to split the processed signal into a low band signal, a mid band signal, and a high band signal,a low band compressor configured to modulate the low band signal to create a modulated low band signal,a high band compressor configured to modulate the high band signal to create a modulated high band signal, anda second processing module configured to process the modulated low band signal, the mid band signal, and the modulated high band signal to create an output signal.2. A system as recited in wherein said first filter module comprises a first low shelf filter configured to ...

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Номер записи: 12
16-01-2020 дата публикации

SIGNAL GAIN CONTROL METHOD AND ELECTRONIC DEVICE

Номер: US20200021261A1
Автор: CHANG Jia-Ren, Tai Kuei-Ting, Yu Ming-Chun
Принадлежит:

A signal gain control method for an electronic device including an alternating-current (AC) impedance detector and a direct-current (DC) impedance detector is provided. An electronic device may implement the method and may be coupled to an external device. The method includes: detecting an AC impedance of the external device by the AC impedance detector; detecting a DC impedance of the external device by the DC impedance detector; obtaining a gain adjustment value according to the AC impedance and the DC impedance; and adjusting a signal gain of an audio signal output to the external device according to the gain adjustment value. 1. A method performed by an electronic device , the method comprising:detecting an alternating current (AC) impedance of an external device by transmitting a frequency sweep signal to the external device and detecting candidate AC impedances responsive to the frequency sweep signal;detecting a direct current (DC) impedance of the external device;determining a gain adjustment value according to the AC impedance and the DC impedance; andadjusting a signal gain of an audio signal output to the external device according to the gain adjustment value.2. (canceled)3. The method of claim 1 , wherein the AC impedance is a maximum of the candidate AC impedances.4. The method of claim 1 , further comprising determining an initial gain adjustment value and adjusting the signal gain of the audio signal output to the external device according to the initial gain adjustment value.5. The method of claim 4 , wherein the initial gain adjustment value is used to adjust the signal gain of the audio signal output to the external device when a difference between the AC impedance and the DC impedance is lower than a predetermined threshold.6. The method of claim 4 , wherein the initial gain adjustment value is trimmed and then used to adjust the signal gain of the audio signal output to the external device when a difference between the AC impedance and the DC ...

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Номер записи: 13
26-01-2017 дата публикации

AUTOMATIC CAR AUDIO VOLUME CONTROL TO AID PASSENGER CONVERSATION

Номер: US20170026764A1
Автор: RAJENDRAN SHANTHA KUMARI
Принадлежит:

A method of controlling the volume level of an audio signal in a vehicle includes the computer-implemented steps of detecting human speech within a passenger compartment of the vehicle, and automatically adjusting, dependent upon the detecting step, a volume level within the passenger compartment of an audible signal being presented by an audio system of the vehicle. 1. A method of controlling the volume level of an audio signal in a vehicle , the method comprising the computer-implemented steps of:detecting human speech within a passenger compartment of the vehicle; andautomatically adjusting, dependent upon the detecting step, a volume level within the passenger compartment of an audible signal being presented by an audio system of the vehicle.2. The method of wherein a microphone is used in detecting the human speech and the audible signal claim 1 , the detecting including removing effects of the audible signal from a microphone signal produced by the microphone.3. The method of claim 2 , further comprising ascertaining whether the microphone signal includes frequency components within a frequency range of human speech.4. The method of wherein the volume level within the passenger compartment of the audible signal is decreased in response to the detecting of the human speech claim 1 , and is subsequently increased in response to detecting that the human speech has ceased.5. The method of wherein the volume level within the passenger compartment of the audible signal is increased in response to detecting that the human speech has ceased to about the same volume level that the audible signal was at before the human speech was detected.6. The method of wherein the detecting and adjusting steps are cyclically repeated.7. The method of wherein the detecting and adjusting steps are cyclically repeated with a frequency of greater than once every three seconds.8. A method of controlling an audio signal in a vehicle claim 1 , the method comprising the computer-implemented ...

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Номер записи: 14
10-02-2022 дата публикации

TWO-STAGE AUDIO GAIN CIRCUIT BASED ON ANALOG-TO-DIGITAL CONVERSION AND AUDIO TERMINAL

Номер: US20220045658A1
Автор: ALPMAN Erkan, DUSTER Jon Sweat, FENG Haigang, TAN Yulin, ZHANG Liuan, Zhang Ning
Принадлежит: Radiawave Technologies Co., Ltd.

Disclosed are a two-stage audio gain circuit based on analog-to-digital conversion and an audio terminal. The two-stage audio gain circuit includes a PGA configured to receive an analog audio signal and perform programmable gain amplification processing on the received analog audio signal; an ADC configured to convert the analog audio signal after the programmable gain amplification processing into a digital audio signal and output the digital audio signal; a first AGC gain unit configured to perform a first AGC processing on the digital audio signal and output a first gain adjustment value to the PGA, for the PGA to perform gain adjustment on the received analog audio signal; and a second AGC gain unit configured to perform a second AGC processing on the digital audio signal and output a second gain adjustment value to the PGA, for the PGA to perform gain adjustment on the received analog audio signal. 1. A two-stage audio gain circuit based on analog-to-digital conversion , comprising:a programmable gain amplifier (PGA) configured to receive an analog audio signal and perform programmable gain amplification processing on the received analog audio signal;an analog-to-digital converter configured to convert the analog audio signal after the programmable gain amplification processing into a digital audio signal and output the digital audio signal;a first automatic gain control (AGC) gain unit configured to perform a first AGC processing on the digital audio signal and output a first gain adjustment value to the PGA, for the PGA to perform gain adjustment on the received analog audio signal according to the first gain adjustment value; anda second AGC gain unit configured to perform a second AGC processing on the digital audio signal and output a second gain adjustment value to the PGA, for the PGA to perform gain adjustment on the received analog audio signal according to the second gain adjustment value.2. The two-stage audio gain circuit based on analog-to-digital ...

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Номер записи: 15
28-01-2021 дата публикации

Content reproducing electronic apparatus and controlling method thereof

Номер: US20210026592A1
Автор: Byeol PARK, Changkun Park, Daewon YOON, Gohwoon JEONG, Heeran LEE, Namsu HA, Sangshin PARK
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A content reproducing electronic apparatus includes a communicator including circuitry, an audio output device, and a processor configured to receive state information indicating a state of an external electronic apparatus from the external electronic apparatus located around the electronic apparatus via the communicator, obtain noise information corresponding to the received state information, and control the audio output device to output an audio signal of a content based on a volume corresponding to the obtained noise information.

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Номер записи: 16
24-01-2019 дата публикации

SOUND PROCESSING DEVICE, SOUND CONTROL DEVICE, AND SOUND CONTROL METHOD

Номер: US20190028077A1
Автор: Ozaki Seigo, UCHIDA Kazuhiro, Watanabe Kazuhiro
Принадлежит:

A sound processing device includes an amplifier, a current detector, and a controller. The amplifier amplifies a sound signal to drive a loudspeaker. The current detector detects a value of an electric current flowing into the amplifier. The controller controls whether to supply power to the amplifier based on a predicted current value and the current detected by the current detector. The predicted current value is predicted in accordance with a volume setting and a level of the sound signal. 1. A sound processing device comprising:a first amplifier that amplifies a sound signal to drive a first loudspeaker;a first current detector that detects a value of a first electric current flowing into the first amplifier; anda controller that:predicts a first predicted current value to flow into the first amplifier in accordance with a volume setting for the first amplifier and a level of the sound signal; andcontrols whether to supply power to the first amplifier based on the first predicted current value and the value of the first electric current.2. The sound processing device according to claim 1 ,wherein the controller:calculates a first threshold in accordance with the first predicted current value;compares the first threshold and the value of the first electric current; andcontrols whether to supply the power to the first amplifier based on a comparison result of the first threshold and the value of the first electric current.3. The sound processing device according to claim 1 ,wherein the controller controls whether to supply the power to the first amplifier, in consideration of a load of the first loudspeaker.4. The sound processing device according to claim 3 ,wherein the controller calculates the load of the first loudspeaker in accordance with a value of the first electric current flowing into the first amplifier when the first amplifier amplifies a known reference signal.5. The sound processing device according to claim 1 ,wherein the controller determines ...

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Номер записи: 17
23-01-2020 дата публикации

TEMPERATURE ADAPTIVE AUDIO AMPLIFIER DEVICE AND CONTROL METHOD THEREOF

Номер: US20200028474A1
Автор: CHIU SHU-YEH, WENG Chung-Hui, WU YI-JHONG
Принадлежит:

A temperature adaptive audio amplifier device includes a digital analog convertor, a gain controller, an amplifier, a temperature sensor and a decision circuit. The digital analog convertor transforms a digital audio signal into an analog convertor. The gain controller includes a gain value and is configured to perform gain processing on the analog audio signal and generate a gained analog audio signal. The amplifier is configured to amplify the gained analog audio signal and generates an amplified analog audio signal. The temperature sensor generates a temperature detect signal according to a junction temperature of the amplifier. The decision circuit receives the temperature detect signal and generates an adaptive gain adjustment signal to the gain controller. The adaptive gain adjustment signal is used to adjust the junction temperature of the amplifier to be within an upper temperature threshold and a lower temperature threshold. 1. A device adapted for coupling to a speaker comprising:a digital and analog converter configured for receiving and converting a digital audio to an analog audio signal;a gain controller coupled to the digital and analog converter and configured to perform gain processing on the analog audio signal and generate a gained analog audio signal;an amplifier coupled to the gain controller and configured to amplify the gained analog audio signal and generate an amplified analog audio signal;a temperature sensor coupled to the amplifier and configured for generating a temperature detection signal based on a junction temperature of the amplifier; anda decision circuit coupled between the temperature sensor and the gain controller and configured for receiving the temperature detection signal transmitted by the temperature sensor and for generating a corresponding gain adjustment signal to the gain controller;wherein the decision circuit configures the gain adjustment signal for adjusting the junction temperature of the amplifier to be below an ...

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Номер записи: 18
01-02-2018 дата публикации

PERCEPTUAL POWER REDUCTION SYSTEM AND METHOD

Номер: US20180034433A1
Автор: JIN Wen, Nigam Vivek P., Polleros Robert
Принадлежит:

A method of reducing power consumed by an audio playback system includes: receiving a signal from a digital audio interface; comparing a sound pressure level (SPL) response of a first frequency band of the signal to a total SPL response of the signal, the first frequency band including frequencies lower than a selected frequency; and attenuating signal components in the first frequency band of the signal based upon the comparison of the SPL response of the first frequency band of the signal to the total SPL response of the signal. 1. A method of reducing power consumption by an audio playback system , comprising:receiving a signal from a digital audio interface;comparing a sound pressure level (SPL) response of a first frequency band of the signal to a total SPL response of the signal, the first frequency band including frequencies lower than a selected frequency; andattenuating signal components in the first frequency band of the signal based upon the comparison of the SPL response of the first frequency band of the signal to the total SPL response of the signal.2. The method of claim 1 , wherein comparing the SPL response of the first frequency band of the signal to the total SPL response of the signal includes:determining a ratio of the SPL response of the first frequency band of the signal to the total SPL response of the signal; andcomparing the determined ratio to a threshold ratio, wherein the signal components in the first frequency band of the signal are attenuated when the determined ratio is less than the threshold ratio.3. The method of claim 2 , wherein determining the ratio of the SPL response of the first frequency band of the signal to the total SPL response of the signal includes:determining a root mean square value of the SPL response of the first frequency band of the signal;determining a root mean square value of the total SPL response of the signal; anddetermining a ratio of the root mean square value of the SPL response of the first frequency ...

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Номер записи: 19
01-02-2018 дата публикации

Lifetime of an Audio Playback Device with Changed Signal Processing Settings

Номер: US20180035224A1
Автор: Little Richard
Принадлежит:

A first lifetime estimate is determined for a first acoustic component of a first audio playback device configured with first signal processing settings. A second audio playback device is configured with second signal processing settings. A second lifetime estimate is determined for a second acoustic component of the second audio playback device configured with the second signal processing settings, where the second lifetime estimate is determined based on first lifetime estimate and where the first acoustic component is substantially similar to the second acoustic component. 1. A method comprising:receiving a first lifetime estimate for a first acoustic component comprising a first transducer of a first playback device configured to play back audio content with first signal processing settings,wherein the first lifetime estimate is determined by mechanically stressing the first acoustic component to failure by operating the first audio playback device at a predetermined volume until the first transducer fails;configuring a second playback device to play back audio content with second signal processing settings, wherein the second playback device has a second acoustic component;receiving one or more statistics associated with movement of the second acoustic component while playing back the audio with the second signal processing settings;determining a second lifetime estimate for the second acoustic component configured with the second signal processing settings without mechanically stressing the second acoustic component to failure,wherein the second lifetime estimate is determined based on (i) applying a lifetime acceleration factor to the first lifetime estimate, wherein the lifetime acceleration factor is based on an indication of cyclic movement of the first acoustic component, (ii) an indication of usage of the second audio playback device, and (iii) the one or more statistics, and wherein the first acoustic component is substantially similar to the second ...

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Номер записи: 20
30-01-2020 дата публикации

SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD, AND PROGRAM

Номер: US20200035212A1
Автор: Asada Kohei, MATSUMOTO Kyosuke, YAMABE YUSHI
Принадлежит: SONY CORPORATION

[Object] To shorten a time until an effect is obtained after start of an adaptive process mode on an ambient sound.

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Номер записи: 21
11-02-2016 дата публикации

AUDIO PROCESSING SYSTEM AND AUDIO PROCESSING METHOD

Номер: US20160043699A1
Автор: Matsumoto Hiroyuki, SAWA Hirotaka, TOKUDA Toshimichi, TSUJI Hisashi, Watanabe Shuichi, YOSHIKUNI Shintaro
Принадлежит: Panasonic Intellectual Property Management Co., Ltd.

An audio processing system, includes: an audio collector that collects audio in a non-directivity state using audio collection elements; an operator receives an input of one or more designation directions for audio emphasis for switching from the non-directivity state to a directivity state; an emphasis processor that generates audio data in the directivity state obtained by performing an emphasis process on the audio data in the designation direction from the audio collector using audio data collected by the audio collector according to the input of the designation direction; a volume adjustor that adjusts volume of the audio data in a directivity state; and an audio output that outputs the audio in a non-directivity state collected by the audio collector or the audio in a directivity state after the volume has been adjusted by the volume adjustor. 1. An audio processing system , comprising:an audio collector that includes a plurality of audio collection elements, and collects audio in a non-directivity state using the audio collection elements;an operator that receives an input of one or more designation directions for audio emphasis for switching from the non-directivity state to a directivity state;an emphasis processor that generates audio data in the directivity state obtained by performing an emphasis process on the audio data in the designation direction from the audio collector using audio data collected by the audio collector according to the input of the designation direction;a volume adjustor that adjusts volume of the audio data in the directivity state generated by the emphasis processor; andan audio output that outputs the audio in the non-directivity state collected by the audio collector or the audio in the directivity state after the volume is adjusted by the volume adjustor.2. The audio processing system of claim 1 ,wherein the operator receives release of the designation direction for switching from the directivity state to the non-directivity ...

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Номер записи: 22
19-02-2015 дата публикации

Amplifier

Номер: US20150049883A1
Автор: Michael Ibrahim
Принадлежит: Mi Audio Pty Limited

An amplifier, such as a vacuum tube guitar amplifier, including a configurable preamplifier, power amplifier and/or power supply. Configuration of sections of the amplifier can be controlled by software, which can be updated directly on the amplifier though an editing interface, by downloading new configuration data from an external device or network, or by updating hardware-based cards or daughter-boards. The amplifier can store multiple circuit designs, and give users access to designs on demand, along with associated potentiometer and switch settings stored in ‘patches’, as well as allow real-time control over potentiometer settings via the amplifier's user interface, or external control via a communications interface. Global settings can also be configured to take into account factors such as type and rating of vacuum tubes. The same methodology can be applied to stand-alone preamplifiers or power amplifiers, solid state amplifiers, preamplifiers or power amplifiers.

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Номер записи: 23
14-02-2019 дата публикации

METHOD FOR AMPLIFYING AUDIO SIGNAL BASED ON ITS AMPLITUDE AND ELECTRONIC DEVICE SUPPORTING THE SAME

Номер: US20190052240A1
Автор: AN Jungyeol, Choi Chulmin, HEO Seungyoon, KEUM Jongmo, KIM Gangyoul, KIM Yangsu, SON Beakkwon
Принадлежит:

An electronic device includes a processor. The processor is configured to examine the sound level of an audio signal obtained from an external object for a preset time; change, if the frequency with which the sound level becomes higher than or equal to a specified level satisfies a preset condition, a reference level, at which the gain value for audio signal amplification is changed in accordance with the amplitude change of the audio signal, for at least one section of the audio signal satisfying the preset condition; and amplify the at least one section of the audio signal according to the changed reference level. 1. An electronic device comprising a processor wherein the processor is configured to:examine a sound level of an audio signal for a preset time, the audio signal obtained from an external source;change, if a frequency with which the sound level becomes higher than or equal to a specified level satisfies a preset condition, a reference level at which a gain value for audio signal amplification is changed in accordance with an amplitude change of the audio signal, for at least one section of the audio signal satisfying the preset condition; andamplify the at least one section of the audio signal according to the changed reference level.2. The electronic device of claim 1 , wherein the processor is configured to:filter the audio signal into a first frequency band signal and a second frequency band signal by using a filter, and for each of the first frequency band signal and second frequency band signals:examine a sound level of the frequency band signal for a preset time;change, if the frequency with which the sound level becomes higher than or equal to a specified level satisfies a preset condition, a reference level at which a gain value for audio signal amplification is changed in accordance with an amplitude change of the frequency band signal, for at least one section of the frequency band signal satisfying the preset condition; andamplify the at ...

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Номер записи: 24
23-02-2017 дата публикации

COMMON-MODE COMPENSATION TECHNIQUE FOR PROGRAMMABLE GAIN AMPLIFIERS

Номер: US20170054425A1
Автор: Azin Meysam, Huang Wenchang, KANAGAL RAMESH Srinidhi Koushik
Принадлежит:

Certain aspects of the present disclosure provide methods and apparatus for adjusting a bandwidth of an amplifier (e.g., a programmable gain amplifier (PGA)). In certain aspects, the PGA generally includes at least one amplification stage having an input and an output, a plurality of compensation capacitors, and at least one first switch configured to selectively couple at least one capacitor of the plurality of compensation capacitors between the input and the output of the at least one amplification stage. In certain aspects, the amplifier includes at least one second switch configured to selectively couple the at least one capacitor to a node such that the at least one capacitor is coupled to only one of the output or the node, where a voltage at the node is a differential mode (DM) reference potential for the amplification stage. 1. A programmable gain amplifier (PGA) , comprising:at least one amplification stage having an input and an output;a plurality of compensation capacitors;at least one first switch configured to selectively couple at least one capacitor of the plurality of compensation capacitors between the input and the output of the at least one amplification stage; andat least one second switch configured to selectively couple the at least one capacitor to a node, wherein the at least one capacitor is coupled to only one of the output or the node and wherein a voltage at the node is a differential mode (DM) reference potential for the at least one amplification stage.2. The PGA of claim 1 , wherein the DM reference potential is a common-mode (CM) voltage of the PGA and wherein the PGA further comprises an amplifier coupled to the node and configured to:compare the CM voltage of the PGA with a CM reference potential; andadjust the CM voltage of the PGA based on the comparison.3. The PGA of claim 1 , wherein the PGA further comprises a common-mode (CM) amplifier coupled to the node and configured to:compare an effective CM voltage of the PGA with a CM ...

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Номер записи: 25
13-02-2020 дата публикации

CAPACITIVE MEMS MICROPHONE WITH ACTIVE COMPRESSION

Номер: US20200052668A1
Автор: Ihs Hassan
Принадлежит:

A digital microphone compresses a large voltage swing signal from a MEMS capacitor to a signal suitable for processing by integrated circuitry. The compression may be performed in an analog domain by selectively coupling adjustment capacitors in parallel to the MEMS capacitor. The digital microphone may decompress the signal in the digital domain using a decompression technique substantially an inverse of the compression performed in the analog domain. 1. A digital microphone , comprising:a microelectromechanical system (MEMS) capacitor;a plurality of adjustment capacitors selectively coupleable in parallel to the MEMS capacitor, the adjustment capacitors selectively coupleable to the MEMS capacitor to achieve a compression in the form of x/(1+N|x|) of a voltage output of the MEMS capacitor, N being a compression index number greater than 1; anda processing chain coupled to one side of the MEMS capacitor, the processing chain including, in sequence, a preamplifier, an analog-to-digital converter (ADC), and a decompression module, the decompression module being configured to perform a decompression substantially an inverse of the compression in the form of x/(1+N|x|).2. The digital microphone of claim 1 , wherein the MEMS capacitor and adjustment capacitors are coupled to a bias voltage greater than a supply voltage coupled to the processing chain.3. The digital microphone of claim 1 , wherein the decompression module is configured to perform the decompression in a digital domain.4. The digital microphone of claim 1 , further comprising a gain error compensation block configured to adjust the compression index based on a ratio of capacitances of the MEMS capacitor and the adjustment capacitors.5. The digital microphone of claim 1 , further comprising a digital filter in the processing chain between the ADC and the decompression module claim 1 , the digital filter configured to suppress quantization noise generated by the ADC.6. The digital microphone of claim 1 , ...

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Номер записи: 26
13-02-2020 дата публикации

CAPACITIVE MEMS MICROPHONE WITH BUILT-IN SELF-TEST

Номер: US20200053496A1
Автор: Ihs Hassan
Принадлежит:

A digital microphone includes built-in self-test features. The features may include capability to apply different bias voltages to a MEMS capacitor sensor of the digital microphone, simulating application of different sound pressures to the digital microphone. The features may also include a digital oscillator, for applying a test signal to an analog front end of the microphone. 1. A method of testing a digital microphone , comprising:applying, at different times, a plurality of different bias voltages to a MEMS capacitor of the digital microphone, the MEMS capacitor serving as a sensor for the digital microphone;measuring outputs of an analog front end of the digital microphone resulting from the application of the plurality of different bias voltages to the MEMS capacitor.2. The method of claim 1 , wherein the different bias voltages are applied using a single bit digital-to-analog converter (DAC).3. The method of claim 2 , wherein the single bit DAC includes a low pass filter.4. The method of claim 1 , wherein an attenuation capacitor is coupled in series with the MEMS capacitor claim 1 , for purposes of creating a gain that is a linear function of electrostatic pressure.5. The method of claim 1 , further comprising applying a digital oscillation signal to an element of the analog front end of the digital microphone claim 1 , and measuring an output of the analog front end.6. The method of claim 5 , wherein the digital microphone includes a preamplifier claim 5 , an analog-to-digital converter (ADC) claim 5 , and a sigma delta block claim 5 , arranged in a sequence.7. The method of claim 6 , wherein the digital oscillation signal is applied as an input to the sigma delta block claim 6 , and the sigma delta block receives a negative feedback signal from the ADC.8. The method of wherein the digital oscillation signal is applied to the element of the analog front end during a test claim 7 , with a signal from the sigma delta block applied to the preamplifier during ...

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Номер записи: 27
10-03-2022 дата публикации

Smart Speaker Power Management

Номер: US20220078548A1
Автор: Dang Zhigang, Lagare Manasi, Naito Daniel, Nussbaum Michael, POON PETER, Sumberg David, Thompson Evan
Принадлежит:

A smart speaker power controller may be configured to receive power conforming to one of a plurality of power modes from an input power source. The power controller may be further configured to communicate the power mode to at least one component of a processing system or at least one component of an audio system for operating the smart speaker according to the received power mode, including determining a current limit of a power converter; determining a shutdown voltage threshold and safely shutting down the smart speaker if the input voltage falls below the threshold, and/or modifying audio operation of the smart speaker in response to the power mode. The smart speaker can further include an audio power supply controller configured to monitor energy required to reproduce an audio signal and to servo a power supply voltage provided to an audio amplifier in response to this energy level. 1. A smart speaker system comprising:a power controller configured to receive power from an input power source, wherein the received power conforms to one of a plurality of power modes;a processing system configured to receive power from the power controller via a processing system power bus; andan audio system configured to receive power from the power controller via an audio system power bus;wherein the power controller is configured to communicate the power mode of the incoming power to at least one component of the processing system or at least one component of the audio system.2. The smart speaker system of wherein the power controller is configured to communicate the power mode to a current limit selector configured to determine a current limit in accordance with the power mode of the incoming power and limit current of a power converter in accordance with the determined current limit.3. The smart speaker system of wherein the power converter is configured to convert input power to power the audio system.4. The smart speaker of wherein the power converter is a boost converter. ...

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Номер записи: 28
02-03-2017 дата публикации

MUSIC REPRODUCING DEVICE

Номер: US20170063319A1
Автор: Kondo Yusuke, TAKEMURA Susumu
Принадлежит:

To provide a music reproducing device that can perform appropriate volume control based on an output. 1. A music reproducing device having multiple volume control comprising: a D/A converter that converts a digital audio signal into an analog audio signal; anda controller, wherein the D/A converter that has volume adjusting function nearest from a headphone output performs volume control related to only a headphone.2. The music reproducing device according to claim 1 ,wherein the controller performs the volume control except the volume control related to only the headphone among the multiple volume control.3. The music reproducing device according to claim 1 ,wherein the volume control related to only the headphone is gain control that controls gain based on impedance of the headphone.4. The music reproducing device according to claim 1 ,wherein the volume control related to only the headphone is sound pressure regulation that limits volume to predetermined sound pressure level.5. The music reproducing device according to claim 1 ,wherein volume control except the volume control related to only the headphone is volume processing that adjusts volume that is received.6. The music reproducing device according to claim 1 ,wherein the volume control except the volume control related to only the headphone is volume maximum setting that limits volume to maximum that is received.7. The music reproducing device according to claim 1 ,wherein the volume control except the volume control related to only the headphone is LR balance processing that adjusts volume balance of L channel and R channel.8. The music reproducing device according to claim 1 , the music reproducing device having a balanced mode and an active control ground mode that amplifies the analog audio signal by one amplifier and maintains ground by the other amplifier of two amplifiers;wherein the volume control except the volume control related to only the headphone is auto volume adjustment that adjusts volume ...

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Номер записи: 29
12-03-2015 дата публикации

METHODS AND SYSTEM FOR WIDEBAND SIGNAL PROCESSING IN COMMUNICATION NETWORK

Номер: US20150071462A1
Автор: Konchitsky Alon
Принадлежит:

The embodiments herein disclose a device and a method for controlling noise in a wideband communication system. In one embodiment herein, multiple microphones for receiving wideband audio signals are provided. A processor is configured to analyze each wideband audio signal received by each microphone. Further, unique signal patterns are generated based on each analyzed wideband signals for each microphone and the unique signal patterns are compared to detect any identical signal patterns. A controller is also provided for controlling gains of those microphones that are detected to be receiving wideband audio signal of identical signal patterns. 1. A wideband noise control communication device comprising:multiple microphones for receiving wideband audio signals; analyze each wideband audio signal received by each microphone,', 'generate unique signal patterns based on each analyzed wideband signals for each microphone, and', 'compare the unique signal patterns to detect any identical signal patterns;, 'a processor configured toa controller for controlling gains of the microphones that are detected to be receiving wideband audio signal of identical signal patterns.2. The communication device of claim 1 , wherein the processor comprises an analog to digital converter for separately converting the wideband analog audio signals received by each microphone claim 1 , into wideband digital signals.3. The communication device of claim 1 , wherein each microphone is placed in a separate position.4. The communication device of claim 1 , wherein the processor analyzes the wideband digital signals and identifies if the detected identical digital signal pattern continually changes its properties.5. The communication device of claim 1 , wherein the controller reduces the gains of the microphones that are detected to be receiving wideband audio signal of identical signal patterns.6. The communication device of claim 1 , wherein the controller does not reduce the gains of the ...

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Номер записи: 30
17-03-2022 дата публикации

INFORMATION HANDLING SYSTEM DYNAMIC SPEAKER VOLUME

Номер: US20220085780A1
Автор: Aurongzeb Deeder M., McKittrick Allen B., Sheu Weijong
Принадлежит: DELL PRODUCTS L.P.

An information handling system housing forms an audio chamber within which a speaker generates audible noise based upon audible information. An adaptive material, such as an electroactive polymer, is disposed at an opening formed in the audio chamber to selectively expand to increase the audio chamber size and contract to decrease the audio chamber size. An increased audio chamber volume enhances audible noise, such as music or audio associated with audiovisual content. A decrease audio chamber can improve airflow within the housing, such as during times of thermal stress. 1. An information handling system comprising:a housing having an audio chamber;a processor disposed in the housing and operable to execute instructions to process information;a memory disposed in the housing and interfaced with the processor, the memory operable to store the instructions and information;a speaker disposed in the audio chamber and interfaced with the processor to apply the information to create audible sounds; andan adaptive material disposed proximate the audio chamber and interfaced with the processor, the adaptive material selectively expanding and contracting based upon one or more predetermined conditions to adjust a volume of the audio chamber.2. The information handling system of wherein the one or more conditions comprise audio presentation of a first or a second audio quality claim 1 , the first audio quality associated with expanding the adaptive material claim 1 , the second audio quality associated with contracting the adaptive material.3. The information handling system of wherein the first audio quality comprises an audiovisual video playback and the second audio quality comprises a notification alert.4. The information handling system of wherein the one or more predetermined conditions comprises a thermal state within the housing claim 1 , the adaptive material contracting above a predetermined temperature threshold.5. The information handling system of wherein the ...

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Номер записи: 31
27-02-2020 дата публикации

AUDIO SIGNAL PROCESSING

Номер: US20200067472A1
Автор: PRIOR Morgan Timothy, RAND Robert
Принадлежит: Cirrus Logic International Semiconductor Ltd.

This application relates to methods and apparatus for audio signal processing, for example by a codec. An audio signal processing module has at least one input for receiving first and second input signals and an output path for outputting an output audio signal. The first and second input signals may be supplied by a first audio component and may correspond to the same source audio data. A controller is configured to compare the first and second input signals and to select one of the first or second input signals as the basis for a signal to be supplied to said output path based on the comparison. The controller may determine a measure of similarity in content between the first and second input signals. 134.-. (canceled)35. An audio signal processing module comprising:at least one input node for receiving first and second input signals;at least one variable gain stage; andan output path for outputting a signal based on a selected one of the first and second input signals; anda controller operable to compare the first and second input signals and control the at least one variable gain stage based on the comparison.36. The audio signal processing module as claimed in wherein the controller is configured to compare said first and second input signals to determine an estimate of any scaling mismatch between the first and second input signals.37. The audio signal processing module as claimed in wherein the controller is configured to condition the estimate of any scaling mismatch by applying at least one of: an attack time constant claim 35 , a delay time constant; a hold-time; a defined ramp rate or range of allowable ramp rates; and hysteresis.38. The audio signal processing module as claimed in wherein the controller is operable to control the at least one variable gain stage to apply a gain to the first input signal based on the estimate of scaling mismatch.39. The audio signal processing module as claimed in wherein the controller is operable to control the at least ...

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Номер записи: 32
19-03-2015 дата публикации

User input with fingerprint sensor

Номер: US20150078586A1
Автор: Diana Dan Jiang, Poon-Keong Ang, Robert Nasry Hasbun
Принадлежит: Amazon Technologies Inc

Devices such as tablets, smartphones, media players, and so forth may incorporate a fingerprint sensor to support acquisition of biometric identification data. As described herein, input data from the fingerprint sensor may be used to control one or more functions of the device. The function controlled may be based at least in part on context of one or more applications executing on the device, direction of motion, and so forth. In one implementation, movement parallel to the fingerprint sensor may modify audio volume settings on the device.

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Номер записи: 33
17-03-2016 дата публикации

SOUND PROCESSING APPARATUS AND SOUND PROCESSING SYSTEM

Номер: US20160080883A1
Автор: Chinen Toru, HATANAKA Mitsuyuki, Shi Runyu, Yamamoto Yuki
Принадлежит: SONY CORPORATION

The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. 1. A sound processing apparatus comprising:at least four sound outputting units;a gain calculating unit configured to determine output gains of sounds to be output from the at least four sound outputting units located close to a sound image localization position as a target position by calculating gains of sounds to be output from the sound outputting units based on a positional relationship between the sound outputting units with respect to each of different combinations among a plurality of combinations of two or three of the at least four sound outputting units, the output gains being to be used for fixing a sound image at the sound image localization position; anda gain adjusting unit configured to perform gain adjustment on sounds to be output from the sound outputting units based on the output gains.2. The sound processing apparatus according to claim 1 , wherein at least four of the output gains each have a value other than 0.3. The sound processing apparatus according to claim 2 , wherein the gain calculating unit includes:a first gain calculating unit configured to calculate output gains of a virtual sound outputting unit and two of the sound outputting units based on a positional relationship among the virtual sound outputting unit, the two of the sound outputting units, and the sound image localization position;a second gain calculating unit configured to calculate gains of other two of the sound outputting units than the two of the sound outputting units based on a positional relationship among the other two of the sound outputting units and the virtual sound outputting unit, the gains of the other two of the sound outputting units being to be used for fixing a sound image at a position of the virtual sound outputting unit; anda calculating unit configured to calculate output gains of the other two of the ...

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Номер записи: 34
05-03-2020 дата публикации

ELECTRONIC DEVICE AND OPERATION METHOD THEREOF

Номер: US20200076389A1
Автор: CHO Jaeyoun, Choi Shukjae, Jeong Hyeonsik, Son Sangmo
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

Provided are an electronic device and an operation method thereof. The operation method of an electronic device for processing an audio signal may include obtaining viewing environment information related to sound intelligibility, processing an input audio signal by separating the input audio signal into a first channel including a primary signal and a second channel including an ambient signal based on the viewing environment information, processing the input audio signal based on a frequency band and based on the viewing environment information, and generating an output signal based on processing the input audio signal. 1. An operation method of an electronic device for processing an audio signal , the operation method comprising:obtaining viewing environment information related to sound intelligibility;processing an input audio signal by separating the input audio signal into a first channel including a primary signal and a second channel including an ambient signal based on the viewing environment information;processing the input audio signal based on a frequency band and based on the viewing environment information; andgenerating an output signal based on processing the input audio signal.2. The operation method of claim 1 , wherein the viewing environment information includes at least one of information associated with ambient noise around the electronic device claim 1 , information associated with a space where the electronic device is located claim 1 , information associated with an ambient device around the electronic device claim 1 , and information associated with an installation environment of the electronic device.3. The operation method of claim 1 , wherein the processing the input audio signal by separating the input audio signal into the first channel including the primary signal and the second channel including the ambient signal comprises:determining weights for the primary signal and the ambient signal based on the viewing environment information; ...

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Номер записи: 35
18-03-2021 дата публикации

METHOD OF MONITORING ELECTRICAL LOADS, CORRESPONDING CIRCUIT, AMPLIFIER AND AUDIO SYSTEM

Номер: US20210083637A1
Автор: Polloni Flavio, Valle Stefano
Принадлежит:

A method of monitoring electrical loads is disclosed. In an embodiment the method includes generating a first voltage signal and a second voltage signal, the second voltage signal in quadrature to the first voltage signal, injecting one of the first voltage signal or the second voltage signal into a signal propagation path towards an electrical load, sensing a current signal flowing through the electrical load as a result of the one of the first voltage signal or the second voltage signal injected into the signal propagation path and processing the first voltage signal, the second voltage signal and the sensed current signal. 1. A method comprising:generating a first voltage signal and a second voltage signal, the second voltage signal in quadrature to the first voltage signal;injecting one of the first voltage signal or the second voltage signal into a signal propagation path towards an electrical load;sensing a current signal flowing through the electrical load as a result of the one of the first voltage signal or the second voltage signal injected into the signal propagation path; and calculating a linear combination of the first voltage signal and the second voltage signal via a first and a second linear combination coefficient,', 'executing an approximation procedure of the sensed current signal via the linear combination, the approximation procedure comprising updating first and second linear combination coefficients until a target approximation is reached, wherein updated values of the first and second linear combination coefficients at the reached target approximation are indicative of an admittance of the electrical load, and', 'calculating the admittance of the electrical load as a function the updated values of the first and second linear combination coefficients at the reached target approximation., 'processing the first voltage signal, the second voltage signal and the sensed current signal, wherein processing comprises2. The method of claim 1 , wherein ...

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Номер записи: 36
18-03-2021 дата публикации

METHOD AND APPARATUS TO EVALUATE AUDIO EQUIPMENT FOR DYNAMIC DISTORTIONS AND OR DIFFERENTIAL PHASE AND OR FREQUENCY MODULATION EFFECTS

Номер: US20210083640A1
Автор: Quan Ronald
Принадлежит:

A system is provided to analyze frequency (or phase) modulation distortion in an audio device, which may include testing with audio frequencies. In one embodiment a change in pitch (or frequency) is measured at an output of the audio device. One or more distortion signals from the audio device may be measured for an amplitude, phase, and or frequency modulation effect. In another embodiment a musical signal may be used as a test signal. Providing additional test signals to the audio device can induce a time varying cross-modulation distortion signal (or static cross-modulation distortion signal) from an output of the audio device. Also utilizing at least one additional filter, filter bank, demodulator and or frequency converter and or frequency multiplier provides extra examination of distortion. Also frequency and or phase response can be measured with the presence of a de-sensing signal and or another signal that induce near slew rate limiting or near overload condition of the device under test. Another system is provided to analyze modulation index differences between input and output signals for a test signal including modulation. 1. An apparatus for measuring a change in a pitch or a frequency from an audio device , wherein the audio device having an input terminal and an output terminal , comprising:coupling to the input of the audio device a test signal comprising a modulated signal wherein the modulated signal includes a first modulating signal, a first carrier signal, and wherein the first carrier signal is modulated by the first modulating signal;the modulated signal having a first pitch variation or a first frequency variation of the first carrier signal, wherein the first pitch variation or the first frequency variation of the first carrier signal is determined by the first modulation signal;coupling an output signal from the output terminal of the audio device to an input terminal of a demodulator to measure the change in the pitch or the change in the ...

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Номер записи: 37
14-03-2019 дата публикации

RECEIVER AND NOISE SQUELCH CONTROL METHOD

Номер: US20190081605A1
Автор: SHIOZAWA Hideaki
Принадлежит:

A noise squelch processor generates a noise squelch determination signal by comparing a noise level and a threshold value with each other. A carrier detector generates a carrier determination signal indicating whether or not a reception signal is present based on a signal strength of the reception signal. An integrator controller controls an integrator to set a cutoff frequency of the integrator to a second cutoff frequency higher than a first cutoff frequency at a first timing when the carrier detector generates a carrier determination signal indicating that the reception signal is present, and to switch the cutoff frequency of the integrator from the second cutoff frequency to the first cutoff frequency at a second timing after an elapse of a predetermined period from the first timing. 1. A receiver comprising:a noise squelch processor configured to generate a noise level decreasing from an initial value to a predetermined convergence value by integrating, by an integrator composed of a low-pass filter, a rectified signal obtained by extracting and rectifying a signal outside a band of a voice in a detection signal, obtained by detecting a reception signal to determine whether a voice signal is to be muted or unmuted by comparing the noise level and a threshold value with each other, and to generate a noise squelch determination signal;a carrier detector configured to generate a carrier determination signal indicating whether or not the reception signal is present based on a signal strength of the reception signal;an audio controller configured to supply a voice signal generated based on the detection signal to a speaker;an audio mute determiner configured to instruct the audio controller to mute the voice signal when the noise squelch determination signal indicates that the voice signal is to be muted; andan integrator controller configured to control the integrator to set a cutoff frequency of the integrator to a second cutoff frequency higher than a first ...

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Номер записи: 38
12-03-2020 дата публикации

PORT VELOCITY LIMITER FOR VENTED BOX LOUDSPEAKERS

Номер: US20200083853A1
Автор: Brunet Pascal M., Li Yuan
Принадлежит:

One embodiment provides a method comprising determining an energy stored in a port of a vented box loudspeaker based on a physical model of the vented box loudspeaker, determining a time-varying attenuation to apply to a source signal for reproduction via the vented box loudspeaker based on the energy stored in the port, and controlling the energy stored in the port in real-time by attenuating the source signal based on the time-varying attenuation. A velocity of air in the port during the reproduction of the source signal is limited based on the controlling. 1. A method comprising:determining an energy stored in a port of a vented box loudspeaker based on a physical model of the vented box loudspeaker;determining a time-varying attenuation to apply to a source signal for reproduction via the vented box loudspeaker based on the energy stored in the port; andcontrolling the energy stored in the port in real-time by attenuating the source signal based on the time-varying attenuation, wherein a velocity of air in the port during the reproduction of the source signal is limited based on the controlling.2. The method of claim 1 , wherein the physical model is a linear model.3. The method of claim 1 , wherein the physical model is a nonlinear model.4. The method of claim 1 , further comprising:recursively determining at least one of an estimated displacement of a diaphragm of an active speaker driver of the vented box loudspeaker, an estimated velocity of the diaphragm, an estimated current flowing through a driver voice coil of the speaker driver, an estimated velocity of the air in the port, and an estimated sound pressure in the vented box loudspeaker based on the physical model of the vented box loudspeaker and a voltage of the source signal.5. The method of claim 1 , wherein the attenuating comprises:determining an instantaneous gain based on the energy stored in the port and a set of parameters;attenuating the gain by applying a smoothing algorithm to the gain; ...

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Номер записи: 39
05-05-2022 дата публикации

COMPENSATION FOR AMBIENT SOUND SIGNALS TO FACILITATE ADJUSTMENT OF AN AUDIO VOLUME

Номер: US20220140798A1
Автор: Ganatra Nitin
Принадлежит:

Embodiments of the invention relate generally to audio device and wearable computing devices to detect and characterize an ambient sound, to adjust an output volume of an audio device. More specifically, disclosed are systems, components and methods to generate audio signals associated with an audio device and an output volume, receive ambient sound signals associated with an ambient sound source, detect the ambient sound signals reaching a threshold intensity, analyzing digital data representing the ambient sound and adjust the audio signals to change the output volume according to a category of the ambient sound signals.

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Номер записи: 40
30-03-2017 дата публикации

SPEECH RECOGNITION SYSTEM AND GAIN SETTING SYSTEM

Номер: US20170092289A1
Автор: Marumoto Toru, Suzuki Hirokazu
Принадлежит: Alpine Electronics, Inc.

When an instruction to start voice input is received from the user, a gain controller acquires, from a gain table which defines a correspondence between vehicle speed ranges and gains, a gain corresponding to a vehicle speed range including the vehicle speed of a vehicle detected by a vehicle speed detector, and sets the acquired gain as the gain of an input amplifier that amplifies an input audio signal output by a microphone. As a gain corresponding to each vehicle speed range, the gain table records a gain of the input amplifier corresponding, in an experimentally determined frequency distribution of peak values in the vehicle speed range, to a maximum frequency in the range of magnitude of voice output as an input audio signal by the microphone and to be input to a speech recognition engine as voice having a magnitude within the input range of the speech recognition engine. 1. A speech recognition system mounted on a vehicle , comprising:a microphone configured to convert input voice into an input audio signal and output the input audio signal;a variable-gain input amplifier configured to amplify the input audio signal output by the microphone;a speech recognition unit configured to perform speech recognition on voice represented by the input audio signal amplified by the input amplifier;a vehicle speed detector configured to detect a vehicle speed of the vehicle;a storage unit configured to store gain control information which records a correspondence between vehicle speeds of the vehicle and gains of the input amplifier; anda gain controller configured to acquire the vehicle speed detected by the vehicle speed detector, and to control a gain of the input amplifier to a gain recorded in the gain control information as a gain corresponding to the acquired vehicle speed.2. The speech recognition system according to claim 1 , wherein the gain control information records a correspondence between each vehicle speed range and a gain of the input amplifier ...

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Номер записи: 41
07-04-2016 дата публикации

SYSTEM FOR REPRODUCING SOUND

Номер: US20160100239A1
Автор: JEONG Hyun-Tae, KIM Bae-Sun, SHIN Hyung-Cheol, SHIN Sung-Yong, SOHN Jong-Moo
Принадлежит:

A system for reproducing a sound in accordance with an embodiment of the present invention includes: a sound reproducing device configured to generate an audio signal corresponding to acoustic data and to allow the audio signal to be carried through a human body; and an earphone configured to output a sound by being vibrated by static electricity generated according to the audio signal carried through the human body. 1. A system for reproducing a sound , comprising:a sound reproducing device configured to generate an audio signal corresponding to acoustic data and to allow the audio signal to be carried through a human body; andan earphone configured to output a sound by being vibrated by static electricity generated according to the audio signal carried through the human body.2. The system of claim 1 , wherein the earphone comprises:a conductor having one end surface thereof formed in a ring shape; andan insulator surrounding an outside of the conductor.3. The system of claim 2 , wherein the conductor is vibrated by an electrostatic attractive force generated between the conductor and a subcutaneous layer of the human body through which the audio signal is carried.4. The system of claim 1 , wherein the sound reproducing device comprises:a sound reproducing part configured to generate the audio signal corresponding to the acoustic data; anda sound transmitting part configured to transmit the audio signal to the human body after increasing a voltage of the audio signal and lowering an intensity of current of the audio signal below a predetermined value.5. The system of claim 4 , wherein the sound transmitting part comprises:an amplifier configured to amplify the voltage of the audio signal;a current limiter configured to lower an intensity of current of the voltage-amplified audio signal below a predetermined value; andan electrode configured to allow the audio signal having the intensity of current lowered below the predetermined value to be carried through the ...

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Номер записи: 42
28-03-2019 дата публикации

HIGH END AUDIO PREAMPLIFIER

Номер: US20190098406A1
Автор: CHAMBON Philippe
Принадлежит: AUDIO TECHNOLOGY SWITZERLAND S.A.

A preamplifier system to adjust the level of an audio signal, said system comprising a transient adjustment line and a steady adjustment line, each receiving the input of the preamplifier system, each adjustment line having an input and an output, said system further comprising a setting device to produce a setting adjustment value, a controller to receive the setting adjustment value and to control the transient adjustment line and the steady adjustment line, a switching module connected an output of the preamplifier system and to the output of the transient adjustment line and output of the steady adjustment line, the transient adjustment line being made of semiconductors and the steady adjustment line being made of a coil with multiple selectable windings. 1. A preamplifier system to adjust the level of an audio signal , said system comprising a transient adjustment line and a steady adjustment line , each receiving the input of the preamplifier system , each adjustment line having an input and an output , said system further comprising a setting device to produce a setting adjustment value , a controller to receive the setting adjustment value and to control the transient adjustment line and the steady adjustment line , a switching module connected an output of the preamplifier system and to the output of the transient adjustment line and output of the steady adjustment line , the transient adjustment line being made of semiconductors and the steady adjustment line being made of a coil with multiple selectable windings , the controller being configured to:detect a change of the setting value;instruct the switching module to select as the output of the preamplifier system, the transient adjustment line output;adjust the level of the audio signal through the transient adjustment line according to the new setting value;detect an absence of change of the setting value;adjust the level of the audio signal through the steady adjustment line according to the new ...

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Номер записи: 43
13-04-2017 дата публикации

TEMPERATURE MONITORING FOR LOUDSPEAKERS

Номер: US20170105068A1
Автор: Lesso John Paul
Принадлежит: Cirrus Logic International Semiconductor Ltd.

This application relates to methods and apparatus for determining the temperature of a voice coil of a loudspeaker (), for instance as part of a system () for protecting the loudspeaker from overheating. The method comprises driving the voice coil with signal components at first and second frequencies, wherein the first frequency (f) is higher than the second frequency (f), and determining first and second indications of current (I) and voltage (V) of the voice coil at said first and second frequencies respectively. The method involves determining an estimated ratio value using the first and second indications of current and voltage, wherein said estimated ratio value corresponds to a ratio between a value based on the resistance of the voice coil and a value based on the inductance of the voice coil. The temperature of the voice coil is then determined based on said estimated ratio value and at least one reference value. An impedance extraction module () may extract values for the impedance at the first and second frequencies respectively (Z, Z). In some embodiments a module () may determine estimated values of the resistance (R) and inductance (L) of the voice coil which are used by temperature estimation block () to determine the temperature. 1. A method of determining the temperature of a voice coil of a loudspeaker comprising:driving the voice coil with signal components at first and second frequencies, wherein the first frequency is higher than the second frequency, and determining first and second indications of current and voltage of the voice coil at said first and second frequencies respectively,determining an estimated ratio value using the first and second indications of current and voltage wherein said estimated ratio value corresponds to a ratio between a value based on the resistance of the voice coil and a value based on the inductance of the voice coil; anddetermining the temperature of the voice coil based on said estimated ratio value and at least ...

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Номер записи: 44
04-04-2019 дата публикации

ADJUSTABLE COMPACT LINE OUTPUT CONVERTER WITH AMPLIFIER TRIGGER

Номер: US20190103845A1
Автор: Dong Gu Wen, Hai Wang, VanSickel Larry
Принадлежит:

A line output converter (LOC) combined with a microprocessor to sense LOC output signal strength and respond to usefully high levels by generating a trigger to turn on an amplifier that is receiving the LOC output signal. A short, first predetermined time is programmed into the microcontroller as a minimum time to sense high signal levels before activating the amplifier, in order to suppress transient signals. A longer, second predetermined time is programmed into the microcontroller as a minimum time to sense low signal levels before deactivating the amplifier. The predetermined boundary between a high and a low signal level is also programmed into the microcontroller and, as with the predetermined times, is established responsive to the characteristics of the particular LOC involved. In an embodiment, the LOC is a compact adjustable LOC located on the same PCB as the microcontroller. 1. A line output converter with an amplifier trigger comprising:a. a line output converter adapted to produce a low-level audio signal output from an audio source input; i. whether the low-level output exceeds a predetermined signal strength for a first predetermined period of time; or', 'ii. the low-level output does not exceed said predetermined signal strength for a second predetermined period of time; and, 'b. a microcontroller in communication with said line output converter in relation to said low-level audio signal output and adapted to determine ifc. wherein said microcontroller is responsive to said determination to supply one of an ON trigger and an OFF trigger, respectively, to an amplifier.2. The line output converter with an amplifier trigger of claim 1 , wherein said first and second predetermined periods of time are programmed into said microprocessor.3. The line output converter with an amplifier trigger of claim 2 , wherein said first and second predetermined periods of time are programmed into said microprocessor responsive to characteristics of said line output ...

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Номер записи: 45
21-04-2016 дата публикации

AUTOMATIC VOLUME CONTROL BASED ON SPEECH RECOGNITION

Номер: US20160111087A1
Автор: SRAIL DOUGLAS A.
Принадлежит:

An entertainment system suitable for use in a vehicle and configured to control an output volume of the system includes a microphone and a processor. The microphone is configured to detect sound in a cabin of the vehicle, and output a microphone signal indicative of the sound. The processor is configured to determine a difference signal based on a difference between the microphone signal and an output signal corresponding to sound being output by the system, determine if the difference signal corresponds to human speech, and decrease the output volume of the system when the difference signal corresponds to human speech. 1. An entertainment system suitable for use in a vehicle , said system configured to control an output volume of the system , said system comprising:a microphone configured to detect sound in a cabin of the vehicle, and output a microphone signal indicative of the sound;a processor configured to determine a difference signal based on a difference between the microphone signal and an output signal corresponding to sound being output by the system, determine if the difference signal corresponds to human speech, and decrease the output volume of the system when the difference signal corresponds to human speech.2. The system in accordance with claim 1 , wherein the processor is further configure to increase the output volume to a previous level a delay time after the difference signal indicates that human speech has stopped.3. The system in accordance with claim 1 , wherein the processor is further configure to determine a difference magnitude of the difference signal if the difference signal does not correspond to human speech claim 1 , and increase the output volume of the system when the difference signal does not correspond to human speech and the difference magnitude is greater than a difference threshold.4. A method for controlling an output volume of an entertainment system suitable for use in a vehicle claim 1 , said method comprising the steps ...

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Номер записи: 46
11-04-2019 дата публикации

ELECTRONIC APPARATUS AND CONTROL METHOD THEREOF

Номер: US20190109573A1
Автор: BAIJAL Anant, CHUNG Hyunjoo, JEONG Hyeon-sik, KO Byeong-seob, SON Sang-mo
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

An electronic apparatus and a method for controlling the electronic apparatus are provided. The electronic apparatus includes an audio processor that outputs an audio including a first audio component and a second audio component and a controller that controls the audio processor to change a ratio of the first audio component to the second audio component according to the changed audio level, in response to receiving an instruction to change a level of the output audio. 1. An apparatus comprising:an audio processor configured to output an audio including a first audio component and a second audio component; anda controller configured to control the audio processor to, in response to receiving an instruction to change a level of the output audio, change a ratio of the first audio component to the second audio component based on the changed audio level and a predetermined condition,wherein the predetermined condition comprises a condition input from a user, a condition received from an outside, or a condition recognized by the apparatus, andwherein the controller is configured to control the audio processor to change the ratio of the first audio component by increasing a proportion of the first audio component compared to the second audio component based on the changed audio level being smaller than a first volume, and increasing a proportion of the second audio component compared to the first audio component based on the changed audio level being larger than a second volume.2. The apparatus according to claim 1 , wherein the controller is further configured to control the audio processor to separate the audio into the first audio component and the second audio component and to adjust a gain of at least one of the first audio component and the second audio component to change the ratio of the first audio component to the second audio component.3. The apparatus according to claim 1 , wherein the controller is further configured to control the audio processor to ...

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Номер записи: 47
02-04-2020 дата публикации

AUDIO SIGNAL PROCESSING

Номер: US20200106411A1
Автор: PRIOR Morgan Timothy, RAND Robert
Принадлежит: Cirrus Logic International Semiconductor Ltd.

This application relates to methods and apparatus for audio signal processing, for example by a codec. An audio signal processing module has at least one input for receiving first and second input signals and an output path for outputting an output audio signal. The first and second input signals may be supplied by a first audio component and may correspond to the same source audio data. A controller is configured to compare the first and second input signals and to select one of the first or second input signals as the basis for a signal to be supplied to said output path based on the comparison. The controller may determine a measure of similarity in content between the first and second input signals. 134.-. (canceled)35. An audio apparatus comprising:an input for receiving a source audio data stream; andfirst and second signal paths for generating first and second audio signals from said source audio data stream and outputting said first and second audio signals, wherein the first audio signal substantially corresponds to the source audio data stream;wherein the second signal path comprises at least one module for varying at least one of: the sample rate; bit resolution; gain; and audio content of the second audio signal; andwherein the audio apparatus is configured to output said first and second audio signals simultaneously.36. The audio apparatus as claimed in further comprising a controller configured to compare the first and second audio signals and to select one of the first and second audio signals as the basis for a signal to be supplied to an output path based on the comparison.37. The audio apparatus as claimed in wherein the controller is configured to select the first audio signal if the first and second audio signals are determined to be similar to one another and to select the second audio signal if the first and second audio signals are determined to be dissimilar to one another.38. The audio apparatus as claimed in wherein the first audio signal ...

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Номер записи: 48
09-06-2022 дата публикации

Apparatus and method for automatic volume control with ambient noise compensation

Номер: US20220182029A1
Автор: Chandra Shekar Ksheerasagar, Jeethendra PORAL, Swaroop Mahadeva
Принадлежит: ARRIS Enterprises LLC

An electronic device and method that automatically adjusts an audio output volume level based on a live environmental acoustic scenario input via a microphone using a machine learning algorithm trained with Human Activity Recognition (HAR). Equipped with such an intelligence the electronic device classifies ambient sounds occurring in the environment of the listening area in which the device is situated into different acoustic scenario mappings such a voice or conversation, for an ambient human conversation detected event, and noise, such as for example a vacuum cleaner or dish washer noise detected event, and automatically adjust the audio output volume accordingly.

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Номер записи: 49
18-04-2019 дата публикации

CALIBRATION OF A DUAL-PATH PULSE WIDTH MODULATION SYSTEM

Номер: US20190115886A1
Автор: Das Tejasvi, Fei Xiaofan, GABORIAU Johann G., Melanson John L., MORTON Alan Mark, VELLANKI Amar, ZHAO XIN, Zhu Lei
Принадлежит: Cirrus Logic International Semiconductor Ltd.

A system may include a digital pulse width modulator subsystem, a first path coupled to an output of the digital pulse width modulator subsystem, a second path coupled to the output of the digital pulse width modulator subsystem and configured to drive a closed-loop analog pulse width modulator, a controller to select between the first path and the second path for processing a signal based on one or more characteristics of the signal, wherein a first gain of the first path and a second gain of the second path are approximately equal at the time of switching selection between the first path and the second path or vice versa, in order to minimize artifacts due to the switching. 1. A system comprising:a digital pulse width modulator subsystem;a first path coupled to an output of the digital pulse width modulator subsystem; anda second path coupled to the output of the digital pulse width modulator subsystem and configured to drive a closed-loop analog pulse width modulator, wherein one of the first path and the second path is selected for processing a signal based on one or more characteristics of the signal;wherein a first gain of the first path and a second gain of the second path are approximately equal at a time of switching selection between the first path and the second path or vice versa, in order to minimize artifacts due to the switching.222.-. (canceled)23. The system of claim 1 , wherein:the first path comprises an open-loop driver stage;the second path comprises a closed-loop driver stage; andthe system further comprises circuitry for selecting between the first path and the second path for processing the signal based on the one or more characteristics of the signal.24. The system of claim 23 , wherein the circuitry comprises a multiplexer.25. The system of claim 23 , wherein the circuitry comprises control circuitry for tri-stating at least one driver stage comprising the open-loop driver stage and the closed-loop driver stage in order to disable the at ...

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Номер записи: 50
25-08-2022 дата публикации

METHOD AND APPARATUS TO EVALUATE AUDIO EQUIPMENT FOR DYNAMIC DISTORTIONS AND OR DIFFERENTIAL PHASE AND OR FREQUENCY MODULATION EFFECTS

Номер: US20220271724A1
Автор: Quan Ronald
Принадлежит:

A system is provided to analyze distortion in an electronic or electromechanical device, which may include testing with one or more modulated signals and/or with one or more demodulators. In one embodiment a change in pitch (or frequency) is measured at an output of the device. One or more signals from a demodulator output may be measured for an amplitude, noise, phase, aliasing, spurious signal, and/or frequency modulation effect. In another embodiment a musical signal may be used as a test signal. Providing additional test signals to the device can induce a cross-modulation distortion signal (or time varying cross-modulation distortion signal) from an output of the device. Also utilizing at least one additional filter, filter bank, demodulator and or frequency converter and or frequency multiplier provides extra examination of distortion. Also frequency and/or phase response can be measured with the presence of a de-sensing signal and or another signal that induce near slew rate limiting or near overload condition of the device under test. Another system is provided to analyze modulation index differences between input and output signals for a test signal including modulation. Another system includes providing a nested or layered modulated signal and/or a nested or layered demodulation apparatus or method. 1. An apparatus for measuring induced spurious signal or induced noise from an audio device , wherein the audio device includes an input terminal and an output terminal , comprising:coupling to the input of the audio device a test signal comprising a modulated signal wherein the modulated signal includes a first modulating signal, a first carrier signal, and wherein the first carrier signal is modulated by the first modulating signal;the modulated signal having a first frequency variation of the first carrier signal, wherein the first frequency variation of the first carrier signal is determined by the first modulating signal,further comprising the modulated ...

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Номер записи: 51
25-08-2022 дата публикации

AUDIO SOURCE AMPLIFICATION WITH SPEAKER PROTECTION FEATURES AND INTERNAL VOLTAGE AND CURRENT SENSING

Номер: US20220272449A1
Автор: Chiu Marshall
Принадлежит:

An apparatus for amplifying an audio source includes a speaker and a chip. The chip includes a processor configured to generate a signal and an amplifier element configured to amplify the signal into an amplified signal. The chip further includes a current monitor configured to monitor the current of the amplified signal prior to the amplified signal being output from the chip to the speaker and a voltage monitor configured to monitor the voltage of the amplified signal prior to the amplified signal being output from the chip to the speaker. The processor of the chip is configured to control a power of the amplified signal output from the chip to the speaker based at least on the current and the voltage. 1. An apparatus , comprising:a speaker; and a processor configured to generate a signal;', 'an amplifier element on the chip, the amplifier element configured to amplify the signal into an amplified signal;', 'a current monitor on the chip, the current monitor configured to monitor the current of the amplified signal prior to the amplified signal being output from the chip to the speaker;', 'a voltage monitor on the chip, the voltage monitor configured to monitor the voltage of the amplified signal prior to the amplified signal being output from the chip to the speaker; and', 'wherein the processor of the chip is configured to control a power of the amplified signal output from the chip to the speaker based at least on the current and the voltage., 'a chip electrically coupled to the speaker by an electrical connection, wherein the chip comprises2. The apparatus of claim 1 , wherein the speaker has a temperature limit claim 1 , and wherein the speaker has a temperature coefficient of resistance claim 1 , wherein the processor of the chip is configured to:determine an expected resistance based on the current and the voltage of the amplified signal;determine an expected temperature based on the expected resistance and the temperature coefficient of resistance; ...

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Номер записи: 52
16-04-2020 дата публикации

GAIN ADJUSTMENT DEVICE, REMOTE CONVERSATION DEVICE, AND GAIN ADJUSTMENT METHOD

Номер: US20200120420A1
Автор: MURAMATSU Mikio, UKAI Satoshi
Принадлежит:

A gain adjustment method includes using a sensor to measure a physical quantity that varies according to a distance, and adjusting a level of an audio signal obtained by a microphone according to a measurement result of the sensor. 1. A gain adjustment device comprising:a sensor configured to measure a physical quantity that varies according to a distance;at least one memory configured to store instructions; and 'adjust a level of an audio signal obtained by a microphone according to a measurement result of the sensor.', 'at least one processor configured to execute the instructions to2. The gain adjustment device according to claim 1 , wherein the physical quantity varies according to a distance between the microphone and a sound source.3. The gain adjustment device according to claim 1 , wherein the sensor includes a plurality of microphones and measures a level ratio in the plurality of microphones as the physical quantity.4. The gain adjustment device according to claim 1 , wherein the at least one processor is further configured to execute the instructions to:extract a predetermined band of an input signal from the microphone.5. The gain adjustment device according to claim 1 ,wherein the sensor is configured to transmit electromagnetic waves and measure a time difference from transmission to reception of the electromagnetic waves as the physical quantity.6. A gain adjustment device comprising:a level adjuster configured to adjust a level of an audio signal obtained by a microphone;a sensor configured to measure a physical quantity that varies according to a distance; anda setter configured to set a gain of the level adjuster according to a measurement result of the sensor.7. The gain adjustment device according to claim 6 , wherein the physical quantity varies according to a distance between the microphone and a sound source.8. The gain adjustment device according to claim 6 , wherein the sensor includes:a plurality of microphones; anda calculator configured ...

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Номер записи: 53
01-09-2022 дата публикации

MONITORING LOUDNESS LEVEL DURING MEDIA REPLACEMENT EVENT USING SHORTER TIME CONSTANT

Номер: US20220276831A1
Автор: Coover Robert, Cremer Markus K., Hodges Todd J., Merchant Shashank, Morris Joshua Ernest
Принадлежит:

In one aspect, an example method includes (i) determining, by a playback device, a first loudness level of a first portion of first media content from a first source while the playback device presents the first media content, with the first portion having a first length; (ii) switching, by the playback device, from presenting the first media content from the first source to presenting second media content from a second source; (iii) based on the switching, determining, by the playback device, second loudness levels of second portions of the first media content while the playback device presents the second media content, with the second portions having a second length that is shorter than the first length; and (iv) while the playback device presents the second media content, adjusting, by the playback device, a volume of the playback device based on one or more of the second loudness levels. 1. A method comprising:switching, by a playback device, from presenting first media content from a first source to presenting second media content from a second source;while the playback device presents the second media content from the second source, receiving, by the playback device, an audio signal corresponding to the first media content;receiving, by the playback device, a loudness level profile corresponding to the first media content; andwhile the playback device presents the second media content, adjusting, by the playback device, a volume of the playback device based on the loudness level profile and the audio signal.2. The method of claim 1 , further comprising aligning the audio signal and the loudness level profile.3. The method of claim 2 , wherein aligning the audio signal and the loudness level profile comprises aligning the audio signal and the loudness level profile using fingerprinting.4. The method of claim 2 , wherein aligning the audio signal and the loudness level profile comprises aligning the audio signal and the loudness level profile using a watermark.5. ...

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Номер записи: 54
02-05-2019 дата публикации

AUDIO SIGNAL PROCESSING

Номер: US20190131946A1
Автор: PRIOR Timothy, RAND Robert
Принадлежит: Cirrus Logic International Semiconductor Ltd.

This application relates to methods and apparatus for audio signal processing, for example by a codec. An audio signal processing module () has at least one input () for receiving first and second input signals (Din, Din) and an output path for outputting an output audio signal (Vout). The first and second input signals may be supplied by a first audio component () and may correspond to the same source audio data. A controller () is configured to compare the first and second input signals and to select one of the first or second input signals as the basis for a signal to be supplied to said output path based on the comparison. The controller may determine a measure of similarity (Sim) in content between the first and second input signals. 134.-. (canceled)35. An audio signal processing module comprising:at least one input node for receiving first and second input signals;an output path for outputting an output audio signal; anda controller configured to compare the first and second input signals and to select one of the first or second input signals as the basis for a signal to be supplied to said output path based on the comparison.36. An audio signal processing module as claimed in wherein the controller is configured to select the first input signal if the first and second input signals are determined to be similar to one another and to select the second input signal if the first and second input signals are determined to be dissimilar to one another.37. An audio signal processing module as claimed in wherein the first input signal corresponds to a higher quality signal for audio content of both of the input signals than the second input signal.38. An audio signal processing module as claimed in wherein the controller is configured to identify one of two received input signals as the first input signal and the other as the second input signal based on at least one of sample rate and digital resolution.39. An audio signal processing module as claimed in wherein ...

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Номер записи: 55
02-05-2019 дата публикации

AUDIO CIRCUIT

Номер: US20190131947A1
Автор: SAKAI Mitsuteru
Принадлежит:

A D/A converter coverts a digital audio signal into an analog audio signal. An analog volume circuit receives an output of the D/A converter. A controller controls the analog volume circuit. The controller shortens transition time at each step in the analog volume circuit further as gain is lower. 1. An audio circuit comprising:a digital/analog (D/A) converter structured to covert a digital audio signal into an analog audio signal;an analog volume circuit coupled to receive an output of the D/A converter; anda controller structured to control gain of the analog volume circuit,wherein the controller is structured to shorten transition time at each step in the analog volume circuit further as the gain is lower.2. The audio circuit according to claim 1 , wherein the controller includes a table showing a relationship between the gain or a range of the gain and transition time corresponding to the gain or the range of the gain.3. The audio circuit according to claim 2 , wherein the controller includes the plurality of tables claim 2 , and the tables are selectable for use.4. The audio circuit according to claim 1 , wherein the controller is implemented in a digital signal processor (DSP) structured to process the digital audio signal.5. An audio circuit comprising:an isolation amplifier coupled to receive an analog audio signal;an analog volume circuit coupled to receive an output signal of the isolation amplifier; anda controller structured to control gain of the analog volume circuit,wherein the controller is structured to shorten transition time at each step in the analog volume circuit further as the gain is lower.6. The audio circuit according to claim 1 , wherein the audio circuit is integrated on one semiconductor substrate.7. An in-vehicle audio device comprising:{'claim-ref': {'@idref': 'CLM-00001', 'claim 1'}, 'the audio circuit according to .'}8. An audio component device comprising:{'claim-ref': {'@idref': 'CLM-00001', 'claim 1'}, 'the audio circuit according ...

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Номер записи: 56
23-04-2020 дата публикации

Amplifying Device and Controlling Method Therefor

Номер: US20200127606A1
Автор: TOGAWA Takeshi
Принадлежит:

An amplifying device includes a first amplifier, a voltage-limiting circuit, a second amplifier, a first controller, and a second controller. The first amplifier is configured to amplify an input signal. The voltage-limiting circuit is configured to limit a voltage of an output signal from the first amplifier to within a limit range. The second amplifier is configured to amplify an output signal from the voltage-limiting circuit. The first controller is configured to control the limit range in accordance with a current supplied to a load from the second amplifier. The second controller is configured to decrease an amplification factor of the first amplifier in a state in which the voltage-limiting circuit limits the voltage of the output signal from the first amplifier. 1. An amplifying device comprising:a first amplifier configured to amplify an input signal;a voltage-limiting circuit configured to limit a voltage of an output signal from the first amplifier to within a limit range;a second amplifier configured to amplify an output signal from the voltage-limiting circuit;a first controller configured to control the limit range in accordance with a current supplied to a load from the second amplifier; anda second controller configured to decrease an amplification factor of the first amplifier in a state in which the voltage-limiting circuit limits the voltage of the output signal from the first amplifier.2. The amplifying device according to claim 1 , wherein the second controller is configured to control an amplification factor of the first amplifier in accordance with a period of time during which the voltage-limiting circuit limits the voltage of the output signal.3. The amplifying device according to claim 1 , wherein the first controller includes:a current detector configured to generate a detection voltage corresponding to the current supplied to the load from the second amplifier;a voltage adjuster configured to adjust the detection voltage;a signal ...

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Номер записи: 57
17-05-2018 дата публикации

MICROPHONE DRIVING DEVICE AND DIGITAL MICROPHONE INCLUDING THE SAME

Номер: US20180138882A1
Автор: CHO Min-Hyung, KIM Yi-Gyeong, YANG Woo Seok
Принадлежит:

The present disclosure relates to a microphone driving device and a digital microphone including the same. A microphone driving device according to an embodiment of the inventive concept includes a voltage-to-current converter, a current-to-voltage converter, an analog-to-digital converter, a digital amplification unit, and a gain controller. The voltage-to-current converter converts an acoustic signal to an output current signal based on a gain control signal. The current-to-voltage converter converts the output current signal to an amplified voltage signal. The analog-to-digital converter converts the amplified voltage signal to a digital signal. The digital amplification unit amplifies the digital signal to an amplified digital signal based on the gain control signal. The gain controller generates a gain control signal. The microphone driving device and the digital microphone including the same according to the inventive concept may have a wide dynamic range and reduce the influence of noise. 1. A microphone driving device comprising:an analog amplification unit configured to amplify an acoustic signal to an amplified voltage signal based on a gain control signal;an analog-to-digital converter configured to convert the amplified voltage signal to a digital signal;a digital amplification unit configured to amplify the digital signal to an amplified digital signal based on the gain control signal; anda gain controller configured to generate the gain control signal,wherein the analog amplification unit comprises:a voltage-to-current converter configured to convert the acoustic signal to an output current signal; anda current-to-voltage converter configured to convert the output current signal to the amplified voltage signal.2. The device of claim 1 , wherein the voltage-to-current converter comprises:a normal voltage-to-current converter configured to convert the acoustic signal to a normal output current signal; andan adjustment voltage-to-current converter ...

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Номер записи: 58
26-05-2016 дата публикации

AUTOMATED AUDIO ADJUSTMENT

Номер: US20160149547A1
Автор: Rider Tomer, Tatourian Igor
Принадлежит:

Various systems and methods for automated audio adjustment are described herein. A processing system for automated audio adjustment may include a monitoring module to obtain contextual data of a listening environment; a user profile module to access a user profile of a listener; and an audio module to adjust an audio output characteristic based on the contextual data and the user profile, the audio output characteristic to be used in a media performance on a media playback device. 1. A processing system for automated audio adjustment , the processing system comprising:a monitoring module to obtain contextual data of a listening environment;a user profile module to access a user profile of a listener; andan audio module to adjust an audio output characteristic based on the contextual data and the user profile, the audio output characteristic to be used in a media performance on a media playback device.2. The system of claim 1 , wherein to obtain the contextual data claim 1 , the monitoring module is to access a health monitor claim 1 , and wherein the contextual data comprises sensor data indicative of a physiological state of the listener.3. The system of claim 2 , wherein the health monitor is integrated into a wearable device worn by the listener.4. The system of claim 1 , wherein to obtain the contextual data claim 1 , the monitoring module is to analyze a video image claim 1 , and wherein the contextual data comprises data indicative of a number of people present in the listening environment claim 1 , the number of people obtained by analyzing the video image.5. The system of claim 1 , wherein the user profile comprises a history of media performances and of listening volumes.6. The system of claim 1 , wherein the user profile module is to modify the user profile based on the contextual data.7. The system of claim 6 , wherein to modify the user profile claim 6 , the user profile module is to use a machine learning process.8. The system of claim 6 , wherein the ...

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Номер записи: 59
15-09-2022 дата публикации

ADAPTIVE ANALOG TO DIGITAL CONVERTER (ADC) MULTIPATH DIGITAL MICROPHONES

Номер: US20220294466A1
Автор: Perrott Michael
Принадлежит:

Exemplary multipath digital microphone described herein can comprise exemplary embodiments of adaptive ADC range multipath digital microphones, which allow low power to be achieved for amplifiers or gain stages, as well as for exemplary adaptive ADCs in exemplary multipath digital microphone arrangements described herein, while still providing a high DR digital microphone systems. Further non-limiting embodiments can comprise an exemplary glitch removal component configured to minimize audible artifacts associated with the change in the gain of the exemplary adaptive ADCs. 1. An apparatus , comprising:a plurality of adaptive analog-to-digital converters (ADCs), configured to generate one of a plurality of digital signals having different scaling factors of an associated sensor signal;an ADC range control component configured to adjust gain of at least one of the plurality of adaptive ADCs based at least in part on a respective signal level threshold being sensed in at least one of the plurality of digital signals having different scaling factors; anda multiplexing component configured to switch from conveying a first digital signal associated with the at least one of plurality of the adaptive ADCs and a second digital signal associated with another of the plurality of the adaptive ADCs.2. The apparatus of claim 1 , wherein the plurality of adaptive ADCs are operatively coupled to an associated sensor comprising at least one of an acoustic sensor claim 1 , an environmental sensor claim 1 , or a motion sensor configured to provide the associated sensor signal.3. The apparatus of claim 1 , wherein the plurality of adaptive ADCs are configured to be gain adjusted by the ADC range control component according to an ADC range control algorithm.4. The apparatus of claim 1 , wherein the plurality of adaptive ADCs are further configured to be placed in a first low-power mode.5. The apparatus of claim 4 , further comprising:a low-power mode adaptive ADC of the plurality of ...

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Номер записи: 60
31-05-2018 дата публикации

Lifetime of an Audio Playback Device with Changed Signal Processing Settings

Номер: US20180152797A1
Автор: Little Richard
Принадлежит:

A first lifetime estimate is determined for a first acoustic component of a first audio playback device configured with first signal processing settings. A second audio playback device is configured with second signal processing settings. A second lifetime estimate is determined for a second acoustic component of the second audio playback device configured with the second signal processing settings, where the second lifetime estimate is determined based on first lifetime estimate and where the first acoustic component is substantially similar to the second acoustic component. 1. A method comprising:determining a first lifetime estimate for a first acoustic component of a first audio playback device configured with first signal processing settings;configuring a second audio playback device with second signal processing settings;determining a second lifetime estimate for a second acoustic component of the second audio playback device configured with the second signal processing settings, wherein the second lifetime estimate is determined based on first lifetime estimate and wherein the first acoustic component is substantially similar to the second acoustic component; andoutputting the second lifetime estimate.2. The method of claim 1 , wherein the first acoustic component is a transducer claim 1 , and wherein determining a first lifetime estimate comprises operating the first audio playback device at predetermined volume until the first transducer fails.3. The method of claim 1 , wherein determining the second lifetime estimate comprises applying a lifetime acceleration factor to the first lifetime estimate claim 1 , wherein the lifetime acceleration factor is based on an indication of cyclic movement of the first acoustic component and the second acoustic component.4. The method of claim 1 , wherein determining a first lifetime estimate for a first acoustic component of a first audio playback device configured with first signal processing settings comprises ...

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Номер записи: 61
14-05-2020 дата публикации

Signal Processing Method, Signal Processing Device And Signal Processing System

Номер: US20200153401A1
Автор: DU Liming, JIANG Songying, Yao Wei, ZHU Ma
Принадлежит: Shanghai Awinic Technology Co., LTD

A signal processing method, device and system are provided. A fitting amplitude of an input signal is acquired by performing fitting on the input signal according to a predetermined fitting model. A determination result about whether to apply a gain on the input signal is determined based on the fitting amplitude, and a gain signal is obtained based on the determination result. A delay signal acquired by delaying the input signal is processed based on the gain signal, to obtain an output signal. With the signal processing device and system, a simple structure and a small calculation amount are achieved, and the signal processing efficiency and the signal processing effect are improved. 1. A signal processing method , comprising:acquiring a fitting amplitude by performing fitting on an input signal according to a fitting model, wherein the fitting model is predetermined;determining, based on the fitting amplitude, whether to apply a gain on the input signal, and acquiring a gain signal based on a determination result;acquiring a delay signal by delaying the input signal by a predetermined time period, wherein the predetermined time period is identical to a time period for obtaining the gain signal; andprocessing the delay signal based on the gain signal, to obtain a gain output signal.2. The method according to claim 1 , wherein the fitting model is predetermined by performing steps of:acquiring a frequency response curve of a driven element; andperforming fitting on the frequency response curve, to establish the fitting model, wherein the fitting model comprises a transfer function for performing fitting on the frequency response curve.3. The method according to claim 1 , wherein the acquiring a fitting amplitude by performing fitting on an input signal according to a fitting model comprises:performing fitting on the input signal with a transfer function in the fitting model to obtain the fitting amplitude, wherein the input signal serves as an input to the transfer ...

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Номер записи: 62
14-05-2020 дата публикации

DYNAMIC TIME-WEIGHTED SYSTEMS AND METHODS FOR MANAGEMENT OF ACOUSTIC EXPOSURE

Номер: US20200153402A1
Автор: Dearaujo Miguel, Lynch John C.
Принадлежит:

Workplace safety is a principal concern in many environments. Protecting user ears from damage due to extended exposure to unacceptably high sound volume serves as an important component to workplace safety. Monitoring a device, such as a phone, utilized by a user often provides an incomplete picture of the sound level presented to the user. As provided herein, monitoring a user's sound exposure on one device may cause the sound level presented to the user from a second device to become limited. Additionally, over time the sound level limits may be adjusted based on the cumulative historic sound exposure. As a result, the user may avoid exposure to unacceptably high sound levels originating from more than one source and/or over an extended period of time. 1. A system , comprising:a processor to input, process, and output electrical signals;a first audio output for outputting sound;a second audio output for outputting sound at a level controllable, at least in part, by the processor;wherein, the processor, upon receiving a first audio level data from the first audio output, determines whether the first audio level data is above a threshold audio level data; andwherein, the processor, upon determining that the first audio level data is above the threshold audio level data, executes an audio limiting action to cause the second audio output to operate at a reduced level audio output level.2. The system of claim 1 , wherein the processor executes the audio limiting action further comprising causing the second audio output and omit execution of the audio limiting action with respect to the first audio output.3. The system of claim 1 , further comprising:a network interface; anda computer data storage accessible to the processor via the network interface; andwherein the processor executes the audio limiting action, further comprising providing a value to the computer data storage that, when read by the second audio output, causes the second audio output to execute the ...

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Номер записи: 63
08-06-2017 дата публикации

AUDIO PROCESSING SYSTEM

Номер: US20170164127A1
Автор: GAUTAMA Temujin
Принадлежит:

An audio processing system for a multi-speaker system having a speaker and a further speaker coupled to the output of an amplifier is described. The audio processing system includes a temperature calculation module configured to determine a voice coil temperature value of at least one of the speaker and the further speaker, each of the speaker and the further speaker having a respective amplifier-output-voltage-to-current transfer function comprising at least one speaker passband; an audio processor having a first input for receiving an audio signal, a second input coupled to the temperature calculation module and an output for outputting a processed audio signal; wherein the audio processor is configured to selectively alter the amplitude of one or more frequencies of the audio signal within the speaker passband of at least one of the speaker and the further speaker in dependence of the voice coil temperature value of at least one of the speaker and the further speaker. 1. An audio processing system for a multi-speaker system having a speaker and a further speaker coupled to the output of an amplifier , each of the speaker and the further speaker having a respective amplifier-output-voltage-to-current transfer function comprising at least one speaker passband , the audio processing system comprising:a temperature calculation module configured to determine a voice coil temperature value of at least one of the speaker and the further speaker;an audio processor having a first input for receiving an audio signal, a second input coupled to the temperature calculation module and an output for outputting a processed audio signal;wherein the audio processor is configured to selectively alter the amplitude of one or more frequencies of the audio signal within the at least one speaker passband of at least one of the speaker and the further speaker in dependence of the voice coil temperature value of at least one of the speaker and the further speaker.2. The audio processing ...

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Номер записи: 64
18-06-2015 дата публикации

Compensation for ambient sound signals to facilitate adjustment of an audio volume

Номер: US20150171813A1
Автор: Nitin Ganatra
Принадлежит: AliphCom LLC

Embodiments of the invention relate generally to audio device and wearable computing devices to detect and characterize an ambient sound, to adjust an output volume of an audio device. More specifically, disclosed are systems, components and methods to generate audio signals associated with an audio device and an output volume, receive ambient sound signals associated with an ambient sound source, detect the ambient sound signals reaching a threshold intensity, analyzing digital data representing the ambient sound and adjust the audio signals to change the output volume according to a category of the ambient sound signals.

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Номер записи: 65
23-05-2019 дата публикации

SOUND REPRODUCTION DEVICE

Номер: US20190158970A1
Автор: Konno Fumiyasu, NAGASAWA SINNOSUKE, OHNO ICHIRO
Принадлежит:

Provided is a sound reproduction device that can accurately adjust a warning sound output from a loudspeaker. The sound reproduction device includes a loudspeaker unit including a loudspeaker and an identifier that identifies a characteristic of the loudspeaker and an amplifier unit that amplifies a signal corresponding to a sound output from the loudspeaker. The amplifier unit includes an amplifier that amplifies the signal input from a sound source and outputs the amplified signal to the loudspeaker, a determination unit, and an amplification degree adjuster. When the loudspeaker unit and the amplifier unit are connected to each other, the determination unit determines the characteristic of the loudspeaker from the identifier of the connected loudspeaker unit. The amplification degree adjuster adjusts the amplification degree of the signal output from the amplifier according to the characteristic of the loudspeaker determined by the determination unit. 1. A sound reproduction device comprising:a loudspeaker unit including a loudspeaker and an identifier that identifies a characteristic of the loudspeaker; andan amplifier unit that amplifies a signal corresponding to a sound to be output from the loudspeaker,wherein the amplifier unit includes:an amplifier that amplifies the signal input from a sound source and outputs the amplified signal to the loudspeaker;a determination unit that determines the characteristic of the loudspeaker using the identifier; andan amplification degree adjuster that adjusts an amplification degree of the signal output from the amplifier, andwhen the loudspeaker unit and the amplifier unit are connected to each other,the determination unit determines the characteristic of the loudspeaker of the connected loudspeaker unit from the identifier of the connected loudspeaker unit, andthe amplification degree adjuster adjusts the amplification degree of the signal to be output from the amplifier according to the characteristic of the loudspeaker ...

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Номер записи: 66
16-06-2016 дата публикации

Method, head unit and computer-readable recording medium for adjusting bluetooth audio volume

Номер: US20160173046A1
Автор: Seung Woo Lee
Принадлежит: Hyundai Motor Co

A head unit includes a microcomputer storing last volume information corresponding to a finally adjusted volume of each of at least one mobile terminal. The microcomputer outputs an audio stream having a volume value adjusted based on the last volume information when a first paired mobile terminal of the at least one mobile terminal transmits the audio stream.

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Номер записи: 67
24-06-2021 дата публикации

Sound Generation Device And Applications

Номер: US20210195335A1
Автор: Margalit Mordehai
Принадлежит:

An audio system includes at least one sound generation device having at least an acoustic transducer; and an electronic integrated circuit. A communication bus is connected to the at least one sound generation device and is configured to communicate a digital signal comprised of one or more audio streams and control signals. A power bus is connected to the at least one sound generation device. The acoustic transducer includes at least one of the following groups: a membrane and an acoustic modulator; or a membrane, acoustic resonator, and acoustic coupler. The electronic integrated circuit is constructed and arranged to receive the digital signal and generate an analog electric signal to operate the acoustic transducer to generate an audio signal in accordance with the control signal. 1. An audio system comprising:at least one sound generation device comprising at least an acoustic transducer and an electronic integrated circuit;a communication bus connected to the at least one sound generation device and configured to communicate a digital signal comprised of one or more audio streams and control signals; anda power bus connected to the at least one sound generation device;wherein the acoustic transducer includes at least one of the following groups: a membrane and an acoustic modulator; or a membrane, acoustic resonator, and acoustic coupler, andwherein the electronic integrated circuit is configured to receive the digital signal and generate an analog electric signal to operate the acoustic transducer to generate an audio signal in accordance with the control signal.2. The audio system according to claim 1 , wherein the electronic circuit includes a wireless transceiver.3. The audio system according to claim 2 , wherein the digital signal is transmitted via a wireless transmission.4. An audio system comprising:at least one acoustic transducer, wherein the acoustic transducer includes at least one a group of: a membrane and an acoustic modulator; or a membrane, an ...

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Номер записи: 68
29-09-2022 дата публикации

Audio signal processing circuit and audio signal processing method

Номер: US20220312113A1
Автор: HUNG Tien-Chiu
Принадлежит:

An audio signal processing circuit includes a filter, a gain control circuit, and a bass boost circuit. The filter filters an audio signal and produces a filtered signal accordingly. The gain control circuit divides the filtered signal in the time domain into a first segment and a second segment that are substantially equal in length, measures a first amplitude of the first segment and a second amplitude of the second segment, and generates a gain based on the first amplitude which is greater than the second amplitude. The bass boost circuit adjusts the amplitudes of the first segment and the second segment according to the gain. 1. An audio signal processing circuit , comprising:a filter for filtering an audio signal to produce a filtered signal;a gain control circuit, coupled to the filter, for dividing the filtered signal in the time domain into a first segment and a second segment that are substantially equal in length, measuring a first amplitude of the first segment and a second amplitude of the second segment, and generating a gain based on the first amplitude which is greater than the second amplitude; anda bass boost circuit, coupled to the filter and the gain control circuit, for adjusting amplitudes of the first segment and the second segment according to the gain.2. The audio signal processing circuit of claim 1 , wherein the gain control circuit performs spectrum analysis on the first segment and the second segment claim 1 , and the first segment and the second segment have identical or similar frequency distributions.3. The audio signal processing circuit of claim 1 , wherein the filter is a first filter claim 1 , the filtered signal is a first filtered signal claim 1 , and the bass boost circuit generates an intermediate signal claim 1 , the audio signal processing circuit further comprising:a second filter for filtering the audio signal to produce a second filtered signal;a non-linear processing circuit, coupled to the second filter, for processing ...

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Номер записи: 69
21-05-2020 дата публикации

ADAPTIVE ANALOG TO DIGITAL CONVERTER (ADC) MULTIPATH DIGITAL MICROPHONES

Номер: US20200162099A1
Автор: Perrott Michael
Принадлежит:

Exemplary multipath digital microphone described herein can comprise exemplary embodiments of adaptive ADC range multipath digital microphones, which allow low power to be achieved for amplifiers or gain stages, as well as for exemplary adaptive ADCs in exemplary multipath digital microphone arrangements described herein, while still providing a high DR digital microphone systems. Further non-limiting embodiments can comprise an exemplary glitch removal component configured to minimize audible artifacts associated with the change in the gain of the exemplary adaptive ADCs. 1. An apparatus , comprising:a plurality of adaptive analog-to-digital converters (ADCs), configured to generate one of a plurality of digital audio signals having different scaling factors of an associated audio signal;an ADC range control component configured to adjust gain of at least one of the plurality of adaptive ADCs based at least in part on a respective sound pressure level threshold being sensed in at least one of the plurality of digital audio signals having different scaling factors;a plurality of gain compensation components, each associated with one of the plurality of digital audio signals having different scaling factors, wherein the ADC range control component is further configured to adjust gain of at least one of the gain compensation components on a continuous basis to compensate for a change in the gain of the at least one of plurality of the adaptive ADCs; anda glitch removal component configured to minimize audible artifacts associated with the change in the gain of the at least one of plurality of the adaptive ADCs.2. The apparatus of claim 1 , further comprising:a multiplexing component configured to switch from conveying a digital audio signal associated with the at least one of plurality of the adaptive ADCs and another digital audio signal associated with another of the plurality of the adaptive ADCs.3. The apparatus of claim 1 , wherein another adaptive ADC of the ...

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Номер записи: 70
22-06-2017 дата публикации

METHOD AND APPARATUS FOR LEVEL CONTROL IN BLENDING AN AUDIO SIGNAL IN AN IN-BAND ON-CHANNEL RADIO SYSTEM

Номер: US20170179899A1
Автор: Olochwoszcz Gabriel, Pahuja Ashwini, Vincelette Scott
Принадлежит:

A method for processing a digital audio broadcast signal includes: separating an analog audio portion and a digital audio portion of the digital audio broadcast signal; determining the loudness of the analog audio portion and the digital audio portion over a first short time interval; using the loudness of the analog and digital audio portions to calculate a short term average gain; determining a long term average gain; converting one of the long term average gain or the short term average gain to dB; if an output has been blended to digital, adjusting a digital gain parameter by a preselected increment to produce a digital gain parameter; if an output has not been blended to digital, setting the digital gain parameter to the short term average gain; providing the digital gain parameter to an audio processor; and repeating the above steps using a second short time interval. 1. A method for processing a digital audio broadcast signal , the method comprising:(a) separating an analog audio portion of the digital audio broadcast signal from a digital audio portion of the digital audio broadcast signal;(b) determining a loudness of the analog audio portion over a first short time interval;(c) determining a loudness of the digital audio portion over the first short time interval;(d) using the loudness of the analog audio portion and the loudness of the digital audio portion to calculate a short term average gain;(e) determining a long term average gain;(f) converting one of the long term average gain or the short term average gain to dB and providing the converted average gain to an audio processor;(g) if an output of a receiver has been blended to digital, adjusting a digital gain parameter toward the long term average gain by a preselected increment to produce an adjusted digital gain parameter;(h) if the output of the receiver has not been blended to digital, setting the digital gain parameter to the short term average gain;(i) providing the digital gain parameter ...

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Номер записи: 71
28-05-2020 дата публикации

AUDIO AMPLIFIER WITH EMBEDDED BUCK CONTROLLER FOR CLASS-G APPLICATION

Номер: US20200169234A1
Автор: Du Ru Feng, LI XiangSheng
Принадлежит:

An audio amplifier includes: a buck controller configured to control an output voltage at a first supply terminal, the output voltage selected from a set including a plurality of output voltages, where the output voltage takes a settling time to settle; a first audio bridge including: a class-AB driver stage coupled to the first supply terminal, and a delay insertion circuit configured to receive a processed digital stream and provide the processed digital stream to the class-AB driver stage a delay time after receiving the processed digital stream, where the delay time is based on the settling time; and an audio amplitude detector configured to detect a first peak amplitude in the first digital audio stream, where the buck controller is configured to select a lowest output voltage from the set that is higher than the first peak amplitude plus a headroom voltage. 1. An audio amplifier comprising:a first supply terminal;a second supply terminal;a buck controller having a supply input configured to receive a battery voltage, the buck controller configured to control an output voltage at the first supply terminal, the output voltage being selected from a set comprising a plurality of output voltages, wherein the output voltage at the first supply terminal takes a settling time to settle when the buck controller changes the output voltage from a first voltage of the set to a second voltage of the set, the second voltage being higher than the first voltage; a class-AB driver stage coupled to the first supply terminal and configured to be coupled to the first speaker,', 'a digital signal processing circuit coupled to the input of the first audio bridge, and', 'a delay insertion circuit configured to receive a processed digital stream from the digital signal processing circuit and configured to provide the processed digital stream to the class-AB driver stage a delay time after receiving the processed digital stream, wherein the delay time is based on the settling time; ...

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Номер записи: 72
28-06-2018 дата публикации

Audio processing apparatus and method of controlling same

Номер: US20180183399A1
Автор: Takatoshi Saeki
Принадлежит: Canon Inc

An audio processing apparatus comprises a signal processing unit configured to process an input audio signal by applying a gain to the input audio signal to generate an audio signal for monitoring by a user. The signal processing unit is configured to increase the gain applied to the input audio signal, in a case where the audio signal for monitoring becomes low, in two stages by first increasing a gain applied to the audio signal at first rate during a first stage and subsequently increasing the gain applied to the audio signal at a second rate during a second stage.

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Номер записи: 73
28-06-2018 дата публикации

Sound processing apparatus and sound processing system

Номер: US20180184222A1
Автор: Chinen Toru, HATANAKA Mitsuyuki, Shi Runyu, Yamamoto Yuki
Принадлежит: SONY CORPORATION

The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. 1. A sound processing apparatus comprising:at least four sound outputting units;a gain calculating unit configured to determine output gains of sounds to be output from the at least four sound outputting units located close to a sound image localization position as a target position by calculating gains of sounds to be output from the sound outputting units based on a positional relationship between the sound outputting units with respect to each of different combinations among a plurality of combinations of two or three of the at least four sound outputting units, the output gains being to be used for fixing a sound image at the sound image localization position; anda gain adjusting unit configured to perform gain adjustment on sounds to be output from the sound outputting units based on the output gains.2. The sound processing apparatus according to claim 1 , wherein at least four of the output gains each have a value other than 0.3. The sound processing apparatus according to claim 2 , wherein the gain calculating unit includes:a first gain calculating unit configured to calculate output gains of a virtual sound outputting unit and two of the sound outputting units based on a positional relationship among the virtual sound outputting unit, the two of the sound outputting units, and the sound image localization position;a second gain calculating unit configured to calculate gains of other two of the sound outputting units than the two of the sound outputting units based on a positional relationship among the other two of the sound outputting units and the virtual sound outputting unit, the gains of the other two of the sound outputting units being to be used for fixing a sound image at a position of the virtual sound outputting unit; anda calculating unit configured to calculate output gains of the other two of the ...

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Номер записи: 74
29-06-2017 дата публикации

MULTI-PATH ANALOG FRONT END AND ANALOG-TO-DIGITAL CONVERTER FOR A SIGNAL PROCESSING SYSTEM WITH LOW-PASS FILTER BETWEEN PATHS

Номер: US20170188149A1
Автор: Melanson John L., SATOSKAR Aniruddha
Принадлежит: Cirrus Logic International Semiconductor Ltd.

In accordance with embodiments of the present disclosure, a processing system comprising may include a plurality of processing paths and a filter. The plurality of processing paths may include a first processing path and a second processing path, wherein the first processing path is configured to generate a first digital signal based on an analog input signal and the second processing path is configured to generate a second digital signal based on the analog input signal. The filter may have a corner frequency and may be configured to generate a filtered digital output signal combining spectral components of the first digital signal lower than the corner frequency and spectral components of the second digital signal higher than the corner frequency to generate a filtered digital signal. 1. A system comprising: the first signal path is configured to generate a first path output signal based on an analog input signal; and', 'the second signal path is configured to generate a second path output signal based on the input signal; and', 'a filter having a corner frequency and configured to generate a filtered output signal combining spectral components of the first path output signal lower than the corner frequency and spectral components of the second path output signal higher than the corner frequency to generate the filtered output signal., 'a plurality of signal paths including a first signal path and a second processing signal path, wherein2. The system of claim 1 , wherein the filter comprises:a low-pass filter configured to low-pass filter a difference between the second path output signal and the first path output signal to generate a filtered difference in accordance with a corner frequency of the low-pass filter; anda combiner configured to sum the second path output signal and the filtered difference to generate the filtered signal, wherein the filtered output signal is based on the filtered signal.3. The system of claim 2 , further comprising a controller ...

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Номер записи: 75
15-07-2021 дата публикации

Methods and Systems for Gain Alignment in Multiple Devices with Reduced Latency

Номер: US20210219054A1
Автор: Chittam Sahiti Priya, Jain Prateek, Kucic Matthew Raymond
Принадлежит:

Methods and systems for aligning amplification gains in a plurality of interconnected devices are disclosed. The method includes receiving by each device limiter gain attenuations and brownout gain attenuations broadcasted by the plurality of devices and selecting the maximum brownout gain attenuation and the maximum limiter gain attenuation. The method includes determining a total attenuation as a sum of the maximum brownout attenuation gain and the maximum limiter attenuation gain. The method includes receiving a frame synchronization signal and adjusting the amplification gain by applying the total attenuation responsive to the frame synchronization signal. 1. A method of aligning amplification gains in a plurality of interconnected devices , the method comprising:receiving by a first device limiter gain attenuations and brownout gain attenuations broadcasted by the plurality of devices;selecting the maximum brownout gain attenuation and the maximum limiter gain attenuation from the broadcasted brownout gain attenuations and limiter gain attenuations;determining a local brownout attenuation based on a supply voltage level and a minimum supply voltage threshold;determining a local limiter attenuation based on the supply voltage level and an input signal amplitude;determining a total attenuation as a sum of the maximum brownout gain attenuation and the maximum limiter gain attenuation; andreceiving by the first device a frame synchronization signal and adjusting the amplification gain by applying the total attenuation responsive to the frame synchronization signal.2. The method of claim 1 , further comprising receiving by the first device the limiter gain attenuations and brownout gain attenuations broadcasted by the plurality of devices in respective time division multiplex (TDM) slots.3. The method of claim 1 , further comprising applying by the first device the total attenuation to the amplification gain upon an occurrence of a rising edge of the frame ...

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Номер записи: 76
16-07-2015 дата публикации

Audio power amplification with reduced input power supply crest factor

Номер: US20150200640A1
Автор: Casey L. Hardy, Joseph J. Curcio, Nathan A. Johanningsmeier, Roderick Hogan
Принадлежит: Apple Inc

A power converter has an output that is coupled in parallel with an energy reservoir circuit and a power supply node of an audio power amplifier. The converter can set an upper limit on its input supply current that is variable in accordance with a control input. A controller is to produce a signal, coupled to the control input of the power converter, that is responsive to a measure of input supply voltage of the power converter and either output voltage of the power converter or output power of the amplifier. Other embodiments are also described and claimed.

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Номер записи: 77
05-07-2018 дата публикации

SOUND ADJUSTMENT DEVICE FOR HEARING PROTECTION AND SOUND ADJUSTMENT METHOD THEREOF

Номер: US20180192193A1
Автор: CHANG Chih-long, Chang Wei-Lin, CHAO Kuan-Li, KO Hsiang-Chun, Li Jian-Ying, YANG Kuo-Ping, YOUNG Neo Bob Chih-Yung
Принадлежит:

A sound adjustment device for hearing protection and a sound adjustment method thereof are disclosed. The method includes: acquiring a sound signal, wherein the sound signal includes a plurality of frequency bands, each frequency band corresponds to a volume; analyzing whether the volume exceeds a setting condition; reducing a gain value of the frequency band which is exceeded the setting condition for reducing the volume of the frequency band and generating an adjusted sound signal; and outputting the adjusted sound signal. 1. A sound adjustment method , used in a sound adjustment device for adjusting a sound signal , the method comprising:acquiring a sound signal, wherein the sound signal comprises a plurality of frequency bands, each frequency band corresponding to a volume;analyzing whether the volume exceeds a setting condition;reducing a gain value of the frequency band which exceeds the setting condition for reducing the volume of the frequency band and generating an adjusted sound signal; andoutputting the adjusted sound signal2. The sound adjustment method as claimed in claim 1 , wherein the volume of the frequency band that did not exceed the setting condition is not reduced.3. The sound adjustment method as claimed in claim 2 , wherein the setting condition includes a volume threshold ranged from 60 dBA to 150 dBA.4. The sound adjustment method as claimed in claim 3 , wherein the setting condition includes a cumulative time threshold ranged from 1 second to 2 hours.5. The sound adjustment method as claimed in claim 4 , wherein when the volume threshold is larger claim 4 , the cumulative time threshold is smaller claim 4 , and when the volume threshold is smaller claim 4 , the cumulative time threshold is larger.6. The sound adjustment method as claimed in claim 4 , wherein the setting condition is the cumulative time threshold that exceeds the volume threshold within a specific time interval.7. The sound adjustment method as claimed in claim 3 , wherein ...

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Номер записи: 78
05-07-2018 дата публикации

HEADPHONE AMPLIFIER CIRCUIT FOR HEADPHONE DRIVER, OPERATION METHOD THEREOF, AND USB INTERFACED HEADPHONE DEVICE USING THE SAME

Номер: US20180192220A1
Автор: CHANG Cheng-Pin, CHUANG Tsung-Peng, LEE Shao-Hsun
Принадлежит:

Provided herein are a headphone amplifier circuit for a headphone driver, an operation method thereof, and a universal serial bus (USB) interfaced headphone device using the same. The output stage of the headphone amplifier circuit is improved to have a differential output structure so as to effectively solve the crosstalk issue and increase the isolation between the left and the right channels such that the audio content presents a spatial perception and a distance perception more specifically. 1. A headphone amplifier circuit for a headphone driver , wherein said headphone driver is configured to output a left channel sound and/or a right channel sound , and said headphone driver comprises a first left channel input node , a second left channel input node , a first right channel input node , a second right channel input node , a left headphone and a right headphone , said headphone amplifier circuit comprising:an audio processing circuit configured to generate a left audio signal and a right audio signal; andat least one phase adjuster coupled to said audio processing circuit to perform phase adjustment on said left audio signal and said right audio signal to generate a first adjusted signal and a second adjusted signal, respectively, wherein said first adjusted signal and said left audio signal have an identical amplitude and opposite phases, and said second adjusted signal and said right audio signal have an identical amplitude and opposite phases.2. The headphone amplifier circuit of claim 1 , wherein said headphone amplifier circuit is disposed in an integrated circuit claim 1 , and said integrated circuit comprises a first left channel output node claim 1 , a second left channel output node claim 1 , a first right channel output node and a second right channel output node so as to be connected to said first left channel input node claim 1 , said second left channel input node claim 1 , said first right channel input node and said second right channel input ...

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Номер записи: 79
20-06-2019 дата публикации

CONTEXT-AWARE LOUDNESS CONTROL

Номер: US20190190477A1
Автор: Travaglini Alessandro
Принадлежит:

Context-aware loudness control of audio content may include choosing from a plurality of loudness level models based on an audio reproduction device, measuring loudness level of the audio content based on the chosen loudness model, processing the real-time loudness measurement of the input audio signal to output real-time loudness level adjustment, processing a momentary loudness measurement of the input audio signal to output a momentary loudness level adjustment, processing a short-term loudness measurement of the input audio signal to output a short-term loudness level adjustment, adjusting the input audio signal based on the real-time, momentary, and short-term loudness level adjustments to output a post-processing input signal, measuring long-term loudness of the post-processing input signal to output a long-term loudness measurement, processing the long-term loudness measurement to output a post-processing level adjustment, and processing the real-time, momentary, short-term, and post-processing level adjustments to output an overall loudness level adjustment. 1. A machine or group of machines for providing real-time context-aware loudness control of audio content , comprising:a loudness meter configured to measure loudness of an input audio signal based on a loudness model from a plurality of loudness models and output a real-time loudness measurement of the input audio signal, the loudness model chosen from the plurality of loudness models based on an audio reproduction device to reproduce an output audio signal derived from the input audio signal; a real-time processor configured to receive and process the real-time loudness measurement of the input audio signal and output a real-time loudness level adjustment;', 'a momentary processor configured to receive and process a momentary loudness measurement of the input audio signal from at least one of the loudness meter or the real-time processor, process the momentary loudness measurement of the input audio ...

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Номер записи: 80
11-06-2020 дата публикации

VOLUME ADJUSTING METHOD AND MOBILE TERMINAL

Номер: US20200186113A1
Автор: Gan Gaoting, Xu Zhaoan, YAN Fenggui, Yang Hai, Zheng Zhiyong
Принадлежит:

The embodiment of the disclosure provides a volume adjusting method and device, a mobile terminal and a storage medium. The method includes: acquiring first audio information corresponding to an audio source when detecting that the audio source of a terminal device is playing; determining a first audio amplitude of the first audio information; determining a second audio amplitude of second audio information when the first audio amplitude meets a preset query condition, wherein the second audio information is played after the first audio information; determining corresponding adjustment information when the second audio amplitude meets a preset adjustment condition; and adjusting a volume of the terminal device according to the adjustment information. The disclosure improves the volume adjustment efficiency and reduces the energy consumption of the terminal device. 1. A volume adjusting method , comprising:acquiring first audio information corresponding to an audio source when detecting that the audio source of a terminal device is playing;determining a first audio amplitude of the first audio information;determining a second audio amplitude of second audio information when the first audio amplitude meets a preset query condition, wherein the second audio information is played after the first audio information;determining corresponding adjustment information when the second audio amplitude meets a preset adjustment condition; andadjusting a volume of the terminal device according to the adjustment information.2. The method according to claim 1 , wherein the first audio amplitude meets the preset query condition comprises:the first audio amplitude being larger than a first preset amplitude threshold;wherein the second audio amplitude meets the preset adjustment condition comprises:the second audio amplitude being smaller than a second preset amplitude threshold.3. The method according to claim 1 , wherein the first audio amplitude meets the preset query condition ...

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Номер записи: 81
11-06-2020 дата публикации

SPEAKER EXCURSION PROTECTION

Номер: US20200186921A1
Автор: Palit Supriyo, Singh Abhishek, VALANARASU Kevin Selva Prasanna
Принадлежит:

An audio circuit includes an amplifier, a voltage sensor, a current sensor, and an excursion control circuit. The voltage sensor is coupled to an output of the amplifier. The current sensor is coupled to the output of the amplifier. The excursion control circuit is coupled to the amplifier, the voltage sensor, and the current sensor. The excursion control circuit includes back electro-magnetic force (EMF) measurement, a back-EMF model, and excursion protection. The back-EMF measurement is to measure back electro-magnetic force of a speaker based on voltage measurements received from the voltage sensor and current measurements received from the current sensor. The back-EMF model is updated based on measurements of the back-EMF and is converted to an excursion model. The excursion protection is to limit amplitude of audio signal provided to the amplifier based on the excursion model of the speaker and amplitude of an audio input signal. 13-. (canceled)4. An audio circuit , comprising: a speaker; and', 'a passive radiator or a bass reflex port;, 'a speaker housing, comprisingan amplifier coupled to the speaker;a voltage sensor coupled to the speaker;a current sensor coupled to the speaker; back electro-magnetic force (EMF) measurement to measure back electro-magnetic force of the speaker based on voltage measurements received from the voltage sensor and current measurements received from the current sensor;', 'a back-EMF model that is updated based on measurements of the back-EMF and is converted to an excursion model; and, 'an excursion control circuit coupled to the amplifier, the voltage sensor, and the current sensor, the excursion control circuit comprising wherein the back-EMF model comprises:', 'a second order filter to model back-EMF of the speaker;, 'excursion protection to limit amplitude of audio signal provided to the amplifier based on the excursion model of the speaker and amplitude of an audio input signal;'}a fourth order filter to model back-EMF of the ...

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Номер записи: 82
13-07-2017 дата публикации

ELECTRONIC APPARATUS AND NOISE REDUCTION METHOD

Номер: US20170200443A1
Автор: KUROSAWA Takeshi
Принадлежит: Alpine Electronics, Inc.

An electronic apparatus with a noise reduction function includes a microphone for inputting an external sound signal inside a vehicle, a DSP for generating a noise reduction signal in order to reduce noise of the external sound signal, a determination unit for determining a driving state of the vehicle, a correction unit for relatively increasing a degree of noise reduction when it is determined that the vehicle is in automatic driving and relatively decreasing a degree of noise reduction when it is determined that the vehicle is in manual driving, and a speaker for outputting the corrected noise reduction signal. 1. An electronic apparatus with a noise reduction function , comprising:an input unit configured to input an external sound signal inside a vehicle;a generation unit configured to generate a noise reduction signal in order to reduce noise of the external sound signal;a determination unit configured to determine a driving state of the vehicle;a correction unit configured to correct the noise reduction signal in order to correct a degree of noise reduction based on a determination result of the determination unit; andan output unit configured to output the noise reduction signal corrected by the correction unit.2. The electronic apparatus according to claim 1 ,wherein the determination unit determines whether the vehicle is in automatic driving and if the determination unit determines that the vehicle is in automatic driving, the correction unit corrects the noise reduction signal at a first degree as a relatively high degree of noise reduction.3. The electronic apparatus according to claim 1 ,wherein when the determination unit determines that the vehicle is not in automatic driving, the correction unit corrects the noise reduction signal at a second degree as a relatively low degree of noise reduction.4. The electronic apparatus according to claim 1 ,wherein the correction unit selects a frequency band of a noise reduction signal to be generated.5. The ...

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Номер записи: 83
18-06-2020 дата публикации

METHOD AND DEVICE FOR ADJUSTING AUDIO SIGNAL, AND AUDIO SYSTEM

Номер: US20200195214A1
Автор: Chen Zhe, MA Guilin
Принадлежит: HARMAN INTERNATIONAL INDUSTRIES, INCORPORATED

A method and a device for adjusting an audio signal, and an audio system are provided. The method includes: obtaining a candidate audio signal (S); obtaining a current noise signal in an environment (S); calculating a first plurality of difference values in loudness between the candidate audio signal and the current noise signal (S); modifying the first plurality of difference values with a plurality of target factors, where the plurality of target factors are obtained based on a plurality of test noise signals and a corresponding plurality of test audio signals (S); and modifying the candidate audio signal with the modified first plurality of difference values to obtain a target audio signal (S). Therefore, loudness loss of the audio signal perceived due to the noise signal can be compensated, and an overcompensation issue can be solved. 1. A method for obtaining a plurality of target factors for adjusting an audio signal , comprising:obtaining a test audio signal and a test noise signal, where the test noise signal has a frequency band covering a preset number of barks;obtaining a plurality of difference values in loudness between the test audio signal and the test noise signal; anddetermining a particular target factor of the plurality of target factors for a particular bark for modifying the plurality of difference values, where the particular target factor ranges from 0 to 1 and is determined based on a predetermined requirement.2. The method according to claim 1 , further comprising:repeating above steps until a first predetermined number of target factors for a first predetermined number of barks are obtained based on a plurality of test noise signals and a corresponding plurality of test audio signals.3. The method according to claim 2 , further comprising:obtaining a second predetermined number of target factors for a second predetermined number of barks, by using a linear interpolation method based on the first predetermined number of target factors and ...

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Номер записи: 84
19-07-2018 дата публикации

AUTOMATIC VOLUME CONTROL FOR LAND MOBILE RADIO

Номер: US20180205354A1
Автор: Barker Christian K., Borer Stephen Joseph, Johnson Joshua Alan, Schiring Marshall Pommier
Принадлежит: E.F. Johnson Company

The present disclosure provides a radio and method for automatically adjusting, in response to ambient noise, the volume setting of a radio. The radio comprises a microphone for receiving ambient audio and generating a microphone signal; a codec for receiving the microphone signal and generating a processor signal representative of the ambient audio; and processor circuitry operable to receive the processor signal, determine an ambient audio level in response to the processor signal, and determine an adjusted radio volume to generate a radio volume control signal, wherein the adjusted radio volume is determined by calculating a difference between a baseline volume level and ambient audio level and adding a current volume level setting, wherein the codec is also operable to receive the radio volume control signal and generate an output signal to adjust the radio volume setting to maintain a net difference between the adjusted radio volume and ambient audio. 1. A land mobile radio for use in a land mobile radio (LMR) system , the land mobile radio configured to automatically adjust a current volume level setting of the land mobile radio in response to ambient audio and comprising:a first circuitry configured to receive a processor input signal based on an ambient audio, and in response to the processor input signal based on the ambient audio, calculate a first amplitude difference between the ambient audio and the current volume level setting and generate a radio volume control signal representative of a value of the current volume level setting based on the first amplitude difference;wherein the first circuitry is configured to provide an upper volume limit and a lower volume limit and to adjust the current volume level setting to maintain the current volume level setting within the upper volume limit and the lower volume limit and generate an output signal to set the value of the current volume level setting of the land mobile radio such that a difference is ...

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Номер записи: 85
06-08-2015 дата публикации

Hearing Aid Devices with Reduced Background and Feedback Noises

Номер: US20150222997A1
Автор: Zhimin Fang
Принадлежит: Individual

A personal sound processing apparatus is disclosed that comprises an earphone; a microphone; an electronic sound processor connected between an automatic gain controller (AGC) and a digital volume controller, an automated feedback tracker operatively connected to the microcontroller, and an audio signal amplifier to amplify audio signals which level is controlled by the digital volume controller. The microcontroller is configured to execute electronic instructions for recognizing at least one of a feedback noise condition and a background noise condition in the audio signals and to send a control signal to the digital volume controller to adjust volume level in response to presence of the at least one of the feedback noise condition and the background noise condition.

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Номер записи: 86
26-07-2018 дата публикации

LOUDSPEAKER PROTECTION SYSTEMS AND METHODS

Номер: US20180213322A1
Автор: NAPOLI Roberto, WILLIAMS Stefan
Принадлежит: Cirrus Logic International Semiconductor Ltd.

A method of thermal protection of a voice coil of a loudspeaker comprises: determining an estimate of a temperature of the voice coil, determining an estimate of a rate of change of the temperature of the voice coil, determining an estimate of power dissipation in the voice coil, and generating a gain control signal for modulating a gain applied to an input signal in a signal path between an input terminal and said loudspeaker. The gain control signal is generated as a function of all of said estimate of the temperature of the voice coil, said estimate of the rate of change of the temperature of the voice coil, and said estimate of the power dissipation in the voice coil. 1. A method of thermal protection of a voice coil of a loudspeaker comprising:determining an estimate of a temperature of the voice coil,determining an estimate of a rate of change of the temperature of the voice coil,determining an estimate of power dissipation in the voice coil,generating a gain control signal for modulating a gain applied to an input signal in a signal path between an input terminal and said loudspeaker,wherein the gain control signal is generated as a function of all of said estimate of the temperature of the voice coil, said estimate of the rate of change of the temperature of the voice coil, and said estimate of the power dissipation in the voice coil.2. A method according to claim 1 , further comprising:based on the estimate of the rate of change of the temperature of the voice coil, setting a threshold power value,based on the threshold power value and on the estimate of the temperature of the voice coil, setting an allowed power value, andbased on the allowed power value and the estimate of power dissipation in the voice coil, generating the gain control signal for modulating the gain applied to the input signal in the signal path between the input terminal and said loudspeaker.3. A method according to claim 2 , comprising claim 2 , when the estimate of the rate of change ...

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Номер записи: 87
11-07-2019 дата публикации

METHODS AND APPARATUS FOR VOLUME ADJUSTMENT

Номер: US20190214954A1
Автор: Coover Robert, Cremer Markus K, Scott Jeffrey, Vartakavi Aneesh
Принадлежит:

Apparatus, systems, articles of manufacture, and methods for volume adjustment are disclosed herein. An example method includes identifying media represented in an audio signal, accessing metadata associated with the media in response to identify the media in the audio signal, determining, based on the metadata, an average volume for the media, and adjusting an output volume of the audio signal based on an average gain value, the average gain value determined based on the average volume for the media. 1. An apparatus to adjust audio volume , the apparatus comprising:an audio signal identifier to identify media represented in an audio signal; access metadata associated with the media in response to identifying the media in the audio signal; and', 'determine, based on the metadata, an average volume for the media; and, 'a metadata accessor toa volume adjuster to adjust an output volume of the audio signal based on an average gain value, the average gain value determined based on the average volume for the media.2. The apparatus of claim 1 , further including a real time audio monitor to determine a difference between an average measured volume of a sample of the audio signal and a desired volume level for a specified timespan claim 1 , wherein the volume adjuster to adjust the volume of the audio signal based on a second gain value claim 1 , the second gain value based on the difference.3. The apparatus of claim 2 , wherein the average gain value is an initial volume adjustment applied to the audio signal and the second gain value is a subsequent volume adjustment applied to the audio signal.4. The apparatus of claim 1 , further including a dynamic range compressor to compress the audio signal when the audio signal identifier is unable to identify media represented in the audio signal.5. The apparatus of claim 1 , further including an audio signal accessor to buffer the audio signal claim 1 , the buffering to cause a delay in outputting the audio signal to provide ...

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Номер записи: 88
12-08-2021 дата публикации

SYSTEM AND METHOD FOR AUTOMATIC ADJUSTMENT OF REFERENCE GAIN

Номер: US20210250009A1
Автор: Lester Michael Ryan, Veselinovic Dusan
Принадлежит:

Systems and methods are provided for automatically adjusting a reference gain of an audio mixer having a reference channel for receiving a far end audio signal from a remote location as a reference signal and a plurality of audio input channels for receiving audio signals captured by a plurality of microphone element. An exemplary method includes determining an echo level in an input audio signal received at a given audio input channel, and automatically determining a gain amount for the reference channel based on the echo level. An exemplary system includes a reference gain adjuster configured to automatically determine a gain amount for the reference channel based on an echo level detected in an input audio signal received at a given audio input channel. 1. A system , comprising:a plurality of microphone elements, each microphone element configured to detect sound and convert the sound to a near end audio signal;at least one audio output device configured to output a far end audio signal comprising sound captured at a remote location;an audio mixer having a plurality of audio input channels for receiving audio signals captured by the plurality of microphone elements and a reference channel for receiving the far end audio signal as a reference signal; anda reference gain adjuster configured to automatically determine a gain amount for the reference channel based on an echo level detected in an input audio signal received at a given audio input channel.2. The system of claim 1 , wherein the reference gain adjuster determines the reference gain amount based on an echo-to-reference ratio calculated using the echo level and a reference level calculated for the reference signal.3. The system of claim 2 , wherein the reference gain adjuster is further configured to determine a coherence level based on the input audio signal and the reference signal claim 2 , and if high coherence is found claim 2 , calculating the echo-to-reference ratio based on the echo level.4. The ...

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Номер записи: 89
12-08-2021 дата публикации

AUDIO AMPLIFIER WITH EMBEDDED BUCK CONTROLLER FOR CLASS-G APPLICATION

Номер: US20210250010A1
Автор: Du Ru Feng, LI XiangSheng
Принадлежит:

An audio amplifier includes: a buck controller configured to control an output voltage at a first supply terminal, the output voltage selected from a set including a plurality of output voltages, where the output voltage takes a settling time to settle; a first audio bridge including: a class-AB driver stage coupled to the first supply terminal, and a delay insertion circuit configured to receive a processed digital stream and provide the processed digital stream to the class-AB driver stage a delay time after receiving the processed digital stream, where the delay time is based on the settling time; and an audio amplitude detector configured to detect a first peak amplitude in the first digital audio stream, where the buck controller is configured to select a lowest output voltage from the set that is higher than the first peak amplitude plus a headroom voltage. 1. A method comprising:receiving a first digital audio stream;detecting a first peak amplitude in the first digital audio stream;selecting an output voltage of a buck converter so that the output voltage is a lowest output voltage from a set of buck output voltages that is higher than the first peak amplitude plus a headroom voltage;converting the first digital audio stream into an analog audio signal; andproviding the analog audio signal to a speaker using a class-AB driver stage a delay time after the output voltage of the buck converter settles, wherein the class-AB driver stage receives power from the buck converter.2. The method of claim 1 , wherein the set comprises three buck output voltages.3. The method of claim 2 , wherein a first output voltage of the set is substantially equal to a battery voltage received by the buck converter.4. The method of claim 1 , wherein the headroom voltage is 2 V or lower.5. The method of claim 1 , further comprising:receiving a second digital audio stream; anddetecting a second peak amplitude in the second digital audio stream, wherein selecting the output voltage of ...

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Номер записи: 90
27-08-2015 дата публикации

Method and apparatus for automatically controlling gain based on sensitivity of microphone in electronic device

Номер: US20150244337A1
Автор: Juhee CHANG, Nakjin CHOI, Sangju Lee, Sunghoon JEONG, Sunghwan Ko, Yoongu Choi
Принадлежит: SAMSUNG ELECTRONICS CO LTD

The present disclosure relates to a method and an apparatus for automatically controlling a gain in an electronic device based on a sensitivity of microphone. The method according to an embodiment of the present disclosure includes outputting a reference audio to a speaker and obtaining a sound signal output by the speaker through a microphone, comparing a parameter of the obtained sound signal with a stored parameter, and adjusting a gain of the microphone based on a result of the comparing.

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Номер записи: 91
25-07-2019 дата публикации

METHOD AND APPARATUS TO EVALUATE AUDIO EQUIPMENT FOR DYNAMIC DISTORTIONS AND OR DIFFERENTIAL PHASE AND OR FREQUENCY MODULATION EFFECTS

Номер: US20190229694A1
Автор: Quan Ronald
Принадлежит:

A system is provided to analyze cross-modulation distortion in audio devices, which may include testing with audio frequencies. One or more distortion signals from the audio device may be measured for an amplitude, phase, and or frequency modulation effect. In another embodiment a musical signal may be used as a test signal. Providing additional test signals to the audio device can induce a time varying cross-modulation distortion signal from an output of the audio device. Also utilizing at least one additional filter, filter bank, demodulator and or frequency converter and or frequency multiplier provides extra examination of distortion. Also frequency and or phase response can be measured with the presence of a de-sensing signal and or another signal that induce near slew rate limiting or near overload condition of the device under test. 1. An apparatus for measuring a de-sensing effect on an amplitude response of an audio device , wherein the audio device having an input terminal and an output terminal , comprising:coupling to the input of the audio device at least two signals including at least a first signal, and a second signal;coupling an output signal from the output terminal of the audio device to an input of an amplitude measuring system to measure the amplitude response of a signal related to the second signal at the output terminal of the audio device and adjusting an amplitude of the first signal to be larger in amplitude of the second signal that provides the de-sensing effect on the second signal.2. The apparatus of wherein the output signal of the audio device causes the audio device to be approaching slew rate limiting.3. The apparatus of wherein the output signal of the audio device causes the audio device to be below slew rate limiting.4. The apparatus of wherein the output signal of the audio device causes the audio device to be substantially at slew rate limiting.5. The apparatus of wherein having the audio device to be substantially at slew ...

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Номер записи: 92
03-09-2015 дата публикации

ANALOG AMPLIFIER FOR RECOVERING ABNORMAL OPERATION OF COMMON MODE FEEDBACK

Номер: US20150249435A1
Автор: Cho Byung-Hak, LEE Jong-Woo, LIM Jae-Hyun
Принадлежит:

An analog amplifier is provided. The analog variable amplifier includes a first amplifier stage configured to amplify a bias current to output a first output voltage and a second output voltage that respectively depend on a magnitude of a first input voltage and a second input voltage, a second amplifier stage configured to receive the first output voltage and the second output voltage of the first amplifier stage as inputs and to amplify the received first output voltage and the second output voltage, and at least one auxiliary bias current source coupled to an electrical connection between the first amplifier stage and the second amplifier stage through which the second amplifier stage receives the first output voltage, and coupled to an electrical connection between the first amplifier stage and the second amplifier stage through which the second amplifier stage receives the second output voltage. 1. An analog variable amplifier comprising:a first amplifier stage configured to amplify a bias current to output a first output voltage and a second output voltage that respectively depend on a magnitude of a first input voltage and a second input voltage;a second amplifier stage configured to receive the first output voltage and the second output voltage of the first amplifier stage as inputs and to amplify the received first output voltage and the second output voltage; andat least one auxiliary bias current source coupled to an electrical connection between the first amplifier stage and the second amplifier stage through which the second amplifier stage receives the first output voltage, and coupled to an electrical connection between the first amplifier stage and the second amplifier stage through which the second amplifier stage receives the second output voltage.2. The analog variable amplifier of claim 1 , wherein the first amplifier stage includes a first output transistor for outputting the first output voltage and a second output transistor for outputting the ...

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Номер записи: 93
16-07-2020 дата публикации

Automatic Gain Control in an Active Noise Reduction (ANR) Signal Flow Path

Номер: US20200227022A1
Автор: Carreras Ricardo F., Cattell Joseph H.
Принадлежит:

The technology described in this document can be embodied in a method that includes receiving an input signal captured by one or more sensors associated with an active noise reduction (ANR) headphone, and determining one or more characteristics of a first portion of the input signal. Based on the one or more characteristics of the first portion of the input signal, a gain of a variable gain amplifier (VGA) disposed in an ANR signal flow path can be adjusted, and accordingly, a set of coefficients for a tunable digital filter disposed in the ANR signal flow path can be selected. The method further includes processing a second portion of the input signal in the ANR signal flow path using the adjusted gain and selected set of coefficients to generate a second output signal for the electroacoustic transducer of the ANR headphone. 125-. (canceled)26. A method comprising:receiving an input signal representing audio captured by a microphone of an active noise reduction (ANR) headphone;processing, by a first compensator, a first frequency range of the input signal to generate a first signal for an acoustic transducer of the ANR headphone, wherein the first signal generated by the first compensator includes a first anti-noise signal configured to reduce a noise signal in the first frequency range;processing, by a second compensator disposed in parallel to the first compensator, a second frequency range of the input signal to generate a second signal for the acoustic transducer, wherein the second signal generated by the second compensator includes a second anti-noise signal configured to reduce a noise signal in the second frequency range,detecting, by one or more processing devices, whether the second signal satisfies a threshold condition; generating a control signal configured to adjust a variable gain amplifier that attenuates the second signal without attenuating the first signal, and', 'generating a combined signal for the acoustic transducer by summing the first ...

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Номер записи: 94
23-07-2020 дата публикации

SIGNAL TRACKING-BASED SUPPLY VOLTAGE GENERATION WITH OVER-BOOSTED VOLTAGE

Номер: US20200235712A1
Автор: EKLUND Eric, King Eric J., Larsen Christian, MAY Jeffrey A.
Принадлежит: Cirrus Logic International Semiconductor Ltd.

A power converter may include an input for receiving an input signal and output for generating an intermediate signal that is a power converted signal from the input signal wherein the intermediate signal is determined based on various parameters of a signal path that utilizes the intermediate signal, wherein the various parameters comprise one or more of the following: a peak output signal of the signal path, energy requested over a period of time by the signal path, available energy from an energy source to the power converter, stored energy at an output of the power converter, and stored energy of a battery for providing electrical energy at the input. 1. A power converter comprising:an input for receiving an input signal; andoutput for generating an intermediate signal that is a power converted signal from the input signal wherein the intermediate signal is determined based on various parameters of a signal path that utilizes the intermediate signal, wherein the various parameters comprise one or more of the following: a peak output signal of the signal path, energy requested over a period of time by the signal path, available energy from an energy source to the power converter, stored energy at an output of the power converter, and stored energy of a battery for providing electrical energy at the input.2. The power converter of claim 1 , wherein the intermediate signal is generated by using one or more of the following: a lookup table claim 1 , piecewise linear calculations in a voltage domain claim 1 , time varying supply changes claim 1 , and parameters transferred from a higher level process limiting power availability or current availability.3. The power converter of claim 1 , wherein the power converter is a boost converter.4. The power converter of claim 1 , wherein the intermediate signal varies with respect to an amount of stored energy required by an output of the signal path and available energy from a power supply for providing electrical energy at ...

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Номер записи: 95
31-08-2017 дата публикации

Tuning of Filters

Номер: US20170250678A1
Автор: Holopainen Jarno Mikael, Isomaki Marko Tapio Oskari, Juntunen Tero-Anssi, Karhu Kimmo Kalervo, Karhu Timo Markus, Kilpelainen Jukka, LUKKARI Veikko Sakari, Raty Tuomo Teuvo Tapani
Принадлежит: NOKIA SOLUTIONS AND NETWORKS OY

There is provided a method, including obtaining information indicating at least one reference characteristic; obtaining input data, the input data relating to the output of the tunable filter; determining, based on the input data, at least one characteristic of the tunable filter; upon detecting that the at least one determined characteristic does not match with the at least one reference characteristic, determining tuning instructions for the tunable filter; and applying the tuning instructions in adjusting the tunable filter. 1. A method for tuning a radio or microwave frequency filter by a control apparatus , comprising:obtaining, by the control apparatus, information indicating at least one reference characteristic;obtaining input data, the input data relating to the output of the tunable filter;determining, based on the input data, at least one characteristic of the tunable filter;upon detecting that the at least one determined characteristic does not match with the at least one reference characteristic, determining tuning instructions for the tunable filter; andapplying the tuning instructions in adjusting the tunable filter.2. The method of claim 1 , wherein the reference characteristic is obtained from a reference filter claim 1 , which represents a filter which is tuned to produce an acceptable reference output.3. The method of claim 1 , wherein determining the at least one characteristic further comprises:determining, based on the input data, a filter transfer function; and determining, as the at least one characteristic, a filter coupling matrix on the basis of a structure of the tunable filter and the filter transfer function.4. The method of claim 3 , further comprising:comparing the determined filter coupling matrix against a reference coupling matrix defined by the at least one reference characteristic; andupon detecting, based on the comparison, that the filter coupling matrix does not match with the reference coupling matrix, determining the tuning ...

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Номер записи: 96
17-09-2015 дата публикации

Information processing apparatus, storage medium having stored therein information processing program, information processing system, and information processing method

Номер: US20150263685A1
Автор: Junichiro Miyatake, Kazutoshi Obana, Kei Yamashita, Kochi Kawai, Kouta Minamizawa, Masahiko Inami, Shoji Masubuchi, Takafumi Aoki
Принадлежит: OBANA Kazutoshi

An information processing apparatus sets an amount of adjustment for adjusting an output in accordance with a user operation and generates a sound signal so that a sound volume of sound to be output from a sound output apparatus changes in accordance with the set amount of adjustment. Then, the information processing apparatus generates a vibration signal so that a magnitude of vibration to be imparted by a vibration apparatus changes in accordance with the set amount of adjustment.

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Номер записи: 97
24-09-2015 дата публикации

ALIGNMENT AND RE-ASSOCIATION OF METADATA FOR MEDIA STREAMS WITHIN A COMPUTING DEVICE

Номер: US20150269255A1
Автор: Bauer Claus, Jiang Wenyu, Radhakrishnan Regunthan
Принадлежит:

Techniques for re-associating dynamic metadata with media data are provided. A media processing system creates, with a first media processing stage, binding information comprising dynamic metadata and a time relationship between the dynamic metadata and media data. The binding information may be derived from the media data. While the first media processing stage delivers the media data to a second media processing stage in a first data path, the first media processing stage passes the binding information to the second media processing stage in a second data path. The media processing system re-associates, with the second media processing stage, the dynamic metadata and the media data using the binding information. 1. A method for re-associating dynamic metadata with media data:creating, with a first media processing stage, binding information comprising dynamic metadata and a time relationship between the dynamic metadata and media data, the binding information being derived from the media data;while the first media processing stage delivers the media data to a second media processing stage in a first data path, passing, by the first media processing stage, the binding information to the second media processing stage in a second data path; andre-associating, with the second media processing stage, the dynamic metadata and the media data using the binding information; deriving, with the first media processing stage, one or more media samples from a first version of the media data;', 'creating the time relationship between the dynamic metadata and the media data as one or more time correspondences between one or more units of the dynamic metadata and the one or more media samples derived from the first version of the media data;', 'storing the one or more media samples in the binding information along with the dynamic metadata and the time relationship; and, 'wherein creating binding information includes regenerating, with the second media processing stage, one or ...

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Номер записи: 98
15-08-2019 дата публикации

AMPLIFIER UNIT FOR OPERATING A PIEZOELECTRIC SOUND TRANSDUCER AND/OR A DYNAMIC SOUND TRANSDUCER, AND A SOUND-GENERATING UNIT

Номер: US20190253026A1
Автор: Bottoni Ferruccio, Haensler Markus, Rusconi Clerici Beltrami Andrea
Принадлежит:

An amplifier unit for operating a piezoelectric loudspeaker or microphone includes an audio amplifier and a detection unit, which is configured to detect whether a sound transducer coupled to the amplifier unit is a piezoelectric sound transducer or a dynamic sound transducer. The amplifier unit is configured in such a way that, after a sound transducer has been coupled, the amplifier unit sends a test signal to the coupled sound transducer. Moreover, a sound-generation unit includes a sound transducer and an amplifier unit, which amplifies an audio signal and feeds it to the sound transducer. 1. An amplifier unit for operating a piezoelectric sound transducer and/or a dynamic sound transducer that becomes coupled to the amplifier unit , which comprises:an audio amplifier,a detection unit connected to the audio amplifier and configured for detecting whether a sound transducer that becomes coupled to the amplifier unit is a piezoelectric sound transducer or a dynamic sound transducer;wherein the detection unit is configured in such a way that, after a sound transducer has become coupled to the amplifier unit, the detection unit sends a test signal to the coupled sound transducer and receives from the coupled sound transducer a response signal from which the detection unit determines whether the coupled sound transducer is a piezoelectric sound transducer or a dynamic sound transducer.2. The amplifier unit as claimed in claim 1 , wherein the audio amplifier is configured for operating one of a piezoelectric sound transducer and a dynamic sound transducer.3. The amplifier unit as in claim 1 , further comprising a cabled interface configured for connecting the sound transducer to the amplifier unit and/or to the detection unit.4. The amplifier unit as claimed in claim 3 , wherein the cabled interface comprises a test line configured for connecting the detection unit to the sound transducer or an audio line configured for connecting the audio amplifier to the coupled ...

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Номер записи: 99
06-08-2020 дата публикации

Power Limiter Configuration for Audio Signals

Номер: US20200252033A1
Автор: Faulstich Aaron
Принадлежит:

Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, multiplying the received audio signal with a power feedback signal to create a product audio signal, wherein the feedback signal comprises a low-pass filtered signal, applying a power amplifier to the product audio signal, and providing the amplified product audio signal as an output signal to a speaker. 1. A method comprising:receiving an audio signal at a feedback compressor circuit;multiplying the received audio signal with a power feedback signal to create a product audio signal, wherein the feedback signal comprises a low-pass filtered signal;applying a power amplifier to the product audio signal; andproviding the amplified product audio signal as an output signal to a speaker.2. The method of claim 1 , comprising:multiplying a current feedback signal and a voltage feedback signal of the amplified product audio signal to create a power feedback signal.3. The method of claim 2 , comprising:providing the power feedback signal to a low-pass filter to create the low-pass filtered signal.4. The method of claim 1 , comprisingperforming a threshold comparison of the low-pass filtered signal to a threshold value to determine whether an output signal power level is exceeding threshold value.5. The method of claim 4 , wherein when the output signal power is exceeding the threshold value claim 4 , subtracting the threshold value from the input signal level.6. The method of claim 4 , comprising:performing a logarithmic conversion of the low-pass filtered signal prior to performing the threshold comparison.7. The method of claim 3 , comprising:performing an antilogarithmic conversion to the power feedback signal prior to multiplying the power feedback signal with the received audio signal.8. An apparatus comprising:a receiver configured to receive an audio signal at a feedback compressor circuit;a processor configured tomultiply the received audio ...

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Номер записи: 100