УСТРОЙСТВО И СПОСОБ ДЛЯ ОПРЕДЕЛЕНИЯ ПОКАЗАТЕЛЯ ДЛЯ ВОСПРИНИМАЕМОГО УРОВНЯ РЕВЕРБЕРАЦИИ, АУДИО ПРОЦЕССОР И СПОСОБ ДЛЯ ОБРАБОТКИ СИГНАЛА

Изобретение относится к обработке аудиосигналов и может быть применено в искусственных ревербераторах. Технический результат - улучшение характеристик воспринимаемого звука. Для этого устройство для определения показателя воспринимаемого уровня реверберации в смешанном сигнале, состоящем из прямой компоненты сигнала и реверберационной компоненты сигнала, содержит процессор модели громкости, содержащий каскад перцепционного фильтра для фильтрации «сухой» компоненты сигнала, реверберационной компоненты сигнала или смешанного сигнала, причем каскад перцепционного фильтра выполнен с возможностью моделирования механизма слухового восприятия объекта для получения фильтрованного прямого сигнала, фильтрованного реверберационного сигнала или фильтрованного смешанного сигнала, блок оценки громкости для оценки первого показателя громкости с использованием фильтрованного прямого сигнала и для оценки второго показателя громкости с использованием фильтрованного реверберационного сигнала или фильтрованного ...

ОЦЕНКА ИЗОЛИРУЮЩИХ КАЧЕСТВ ОБТУРАТОРА ДЛЯ УШНОГО КАНАЛА

Изобретение относится к акустике, в частности к средствам измерения звукоизолирующих свойств обтураторов. Устройство содержит микрофон ушного канала, приемник, контур предварительной обработки, блок преобразования частоты, блок оценки качества изоляции. При этом определение показателя качества изоляции выполняется на основе Фурье-анализа с получением частотного спектра сигналов. Затем выполняется сравнение уровней сигнала в первом и втором частотных диапазонах. Показание качества изоляции вычисляется при обнаружении усиления низких частот, либо при изменении величины частотного спектра, либо при вычислении градиента измеренной величины как функции частоты. При этом в качестве тестовых сигналов используют звуки, издаваемые телом человека. В случае несоответствия измеренных характеристик установленным требованиям, формируется сигнал оповещения. Технический результат – снижение габаритных размеров устройства, повышение энергосбережения, повышение точности оценки показателей качества звукоизоляции ...

АКУСТИКО-ЭМИССИОННЫЕ ДАТЧИКИ СО ВСТРОЕННЫМИ АКУСТИЧЕСКИМИ ГЕНЕРАТОРАМИ

Использование: для неразрушающего акустико-эмиссионного контроля. Сущность изобретения заключается в том, что устройство акустико-эмиссионного датчика со встроенным акустическим генератором, содержит акустический приемник; акустический генератор, расположенный рядом с акустическим приемником; корпус, акустический генератор и акустический приемник, расположенные в корпусе; закрепляющий состав в корпусе, чтобы по меньшей мере частично герметизировать акустический генератор и акустический приемник; и износостойкую пластину, находящуюся в акустической связи с акустическим приемником и с акустическим генератором, при этом износостойкая пластина выполнена с возможностью передачи акустической энергии во время испытания от акустического генератора к акустическому приемнику через конструкцию, с которой соединена износостойкая пластина, и при этом износостойкая пластина содержит первую акустическую изоляцию, чтобы препятствовать передаче акустической энергии от акустического генератора к акустическому ...

МОДЕЛЬ УХА, ИСКУССТВЕННАЯ ГОЛОВА И ИЗМЕРИТЕЛЬНАЯ СИСТЕМА И СПОСОБ ИЗМЕРЕНИЯ С ИСПОЛЬЗОВАНИЕМ МОДЕЛИ УХА И ИСКУССТВЕННОЙ ГОЛОВЫ

Изобретение относится к измерительной технике, в частности к акустической метрологии. Измерительная система для оценивания акустического устройства, причем акустическое устройство позволяет пользователю слышать звук посредством вибрации вибрирующего элемента, система содержит модель уха, включающую в себя искусственное ухо, смоделированное в соответствии с ухом человека, и искусственный наружный слуховой проход, составляющий единое целое с искусственным ухом, и микрофон. Благодаря тому, что акустическое устройство находится в состоянии контакта с искусственным ухом, вибрация передается от искусственного уха к искусственному наружному слуховому проходу. Причем твердость по Шору искусственного наружного слухового прохода составляет от 20 до 60. При этом микрофон располагается на конце искусственного наружного слухового прохода. Искусственный наружный слуховой проход присоединен к трубчатому участку. Модель уха присоединена к модели головы. Модель уха включает в себя участок, выполненный из ...

УСТРОЙСТВО ДЛЯ ИСПЫТАНИЙ СРЕДСТВ КОЛЛЕКТИВНОЙ ШУМОЗАЩИТЫ

Полезная модель относится к приборам и устройствам для испытаний средств коллективной защиты от авиационного шума. Устройство для испытаний средств коллективной шумозащиты выполнено в корпусе, имеющем форму куба со скругленными углами, внутри которого жестко закреплены микропроцессор, блок хранения информации с энергонезависимой памятью и аккумулятор, соединенный с разъемом для его подзарядки, встроенным в боковую грань корпуса, в верхнюю грань заподлицо ее поверхности встроены датчик температуры, датчик освещенности, датчик влажности воздуха, индикатор исправности устройства, две противоположные боковые грани оборудованы разъемами для подключения аудиометров и оборудованы креплениями ручки для переноски устройства, каждая из двух других противоположных боковых граней оборудована тремя USB-разъемами, поверхность нижней грани корпуса выполнена ребристой, причем блок хранения информации с энергонезависимой памятью, аккумулятор, датчик температуры, датчик освещенности, датчик влажности воздуха ...

Устройство дистанционного измерения коэффициента преобразования пьезоэлектрического акселерометра в рабочем режиме

Полезная модель относится к области измерительной техники и предназначена для периодического дистанционного измерения коэффициента преобразования пьезоэлектрического акселерометра (далее - пьезоакселерометра) путем возбуждения свободных колебаний его чувствительного элемента и определения их параметров без демонтажа пьезоакселерометра с объекта эксплуатации и без вывода объекта эксплуатации из рабочего режима.Устройство содержит разъем для подключения пьезоакселерометра, переключатель, источник тестового напряжения, усилитель заряда, аналого-цифровой преобразователь, блок измерения и контроля, блок коррекции коэффициента передачи, орган управления, интерфейсный блок, блок аппроксимации и блок определения исправности.Устройство позволяет дистанционно определить текущее значение коэффициента преобразования пьезоакселерометра с целью оперативной коррекции коэффициента передачи измерительного канала системы вибромониторинга. Основным преимуществом является повышенная точность определения коэффициента преобразования за счет снижения чувствительности к импульсным воздействиям и непериодическим помехам. 3 ил. РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 204 614 U1 (51) МПК G01H 17/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ПОЛЕЗНОЙ МОДЕЛИ К ПАТЕНТУ (52) СПК H04R 29/00 (2021.02) (21)(22) Заявка: 2021101996, 28.01.2021 (24) Дата начала отсчета срока действия патента: Дата регистрации: 01.06.2021 (45) Опубликовано: 01.06.2021 Бюл. № 16 2 0 4 6 1 4 R U (56) Список документов, цитированных в отчете о поиске: RU 99158 U1, 10.11.2010. RU 2271015 C1, 27.02.2006. RU 2176396 C1, 27.11.2001. RU 2701207 C1, 25.09.2019. RU 99182 U1, 10.11.2010. US 2017227381 A1, 10.08.2017. (54) Устройство дистанционного измерения коэффициента преобразования пьезоэлектрического акселерометра в рабочем режиме (57) Реферат: Полезная модель относится к области контроля, блок коррекции коэффициента измерительной техники и предназначена для передачи, орган управления, ...

Способ прогнозирования технического состояния и оперативной оценки остаточного ресурса гидроакустического комплекса корабля с применением информационно-сопроводительной системы

Изобретение относится к метрологии. Согласно способу на этапе технического проекта производят расчет надежности с использованием интенсивности отказов, вероятности безотказной работы, время жизни. Затем от системы диагностики и контроля технического состояния принимают информацию о техническом состоянии приборов. При выявлении отказа система диагностики передает информацию о фиксации и локализации в блок сбора информации, в котором происходит аккумулирование информации об отказе. Данная информация поступает в блок статистической обработки. Данная информация копится и передается для выполнения корреляционного анализа, в результате которого выявляется взаимосвязь между выходами из строя конкретных радиоэлементов и диагностическими параметрами, по которым был зафиксирован отказ. От блока статистической обработки информация об отказах поступает на блок перерасчета показателей надежности, в котором происходит перерасчет априорных данных по надежности от блока путем прямого расчета интенсивности ...

СПОСОБ ОДНОВРЕМЕННОГО ИЗМЕРЕНИЯ РАЗБОРЧИВОСТИ РЕЧИ НЕСКОЛЬКИХ ИСТОЧНИКОВ

Изобретение относится к акустической метрологии, в частности к приборам для определения разборчивости речи. Согласно способу прокладывают оптические волокно, разделяют его на измерительные участки, подключают его к измерительному модулю, излучают сигнал, по максимальному времени возврата отраженного сигнала определяют длину волокна, вычисляют период Т посылки измерительных импульсных сигналов. Приемником принимают акустически модулированную последовательность отраженных измерительных импульсов, демодулируют акустическую составляющую принятых отраженных измерительных импульсных сигналов, измеряют уровень естественного шума. Затем размещают М источников сигналов, последовательно каждым из М источников излучают N частот акустических сигналов. Причем частоты излучения распределены по средним частотам N полос, принимают сигналы, вычисляют соотношение сигнал/шум, строят вариационный ряд значений соотношений сигнал/шум, выбирают старший член вариационного ряда, выделяют и запоминают характер изменения ...

УСТРОЙСТВО АДАПТЕРА АУДИОСИГНАЛА, СИСТЕМА ОБНАРУЖЕНИЯ ИНТЕРФЕЙСА АДАПТЕРА АУДИОСИГНАЛА И МАРКЕР ЭЛЕКТРОННОЙ ПОДПИСИ

Изобретение относится к области акустики, в частности к средствам адаптации звуковых сигналов. Адаптер содержит первый и второй интерфейсы, усилитель аудиосигнала и контур полного сопротивления. Первый интерфейс состоит из контакта приема аудиосигнала, контакта микрофона и контакта заземления. Второй интерфейс имеет выходной контакт аудиосигнала. Контур усиления аудиосигнала настроен на усиление аудиосигнала, принимаемого контактом приема аудиосигнала. Аудиосигнал, принимаемый контактом приема аудиосигнала, поступает на контакт микрофона или контакт заземления через эквивалентное сопротивление контура усиления аудиосигнала. Контур сопротивления соединяет контакт микрофона и контакт заземления. Также система содержит средство обнаружения интерфейса, которое имеет собственный интерфейс. Интерфейс устройства обнаружения содержит выходной контакт сигнала обнаружения, контакт заземления и входной контакт обнаружения. Выводной контакт обнаружения подключен к первому интерфейсу адаптера, контакт ...

ОПТИМИЗАЦИЯ ГРОМКОСТИ И ДИНАМИЧЕСКОГО ДИАПАЗОНА ЧЕРЕЗ РАЗЛИЧНЫЕ УСТРОЙСТВА ВОСПРОИЗВЕДЕНИЯ

ОПТИМИЗАЦИЯ ГРОМКОСТИ И ДИНАМИЧЕСКОГО ДИАПАЗОНА ЧЕРЕЗ РАЗЛИЧНЫЕ УСТРОЙСТВА ВОСПРОИЗВЕДЕНИЯ

АКУСТИКО-ЭМИССИОННЫЕ ДАТЧИКИ СО ВСТРОЕННЫМИ АКУСТИЧЕСКИМИ ГЕНЕРАТОРАМИ

УСТРОЙСТВО, СОДЕРЖАЩЕЕ МНОЖЕСТВО АУДИОДАТЧИКОВ, И СПОСОБ ЕГО ЭКСПЛУАТАЦИИ

... 1. Способ эксплуатации устройства, причем устройство содержит множество аудиодатчиков и конфигурировано таким образом, что, когда первый аудиодатчик из множества аудиодатчиков контактирует с пользователем устройства, второй аудиодатчик из множества аудиодатчиков контактирует с воздухом, причем способ содержит этапы, на которых:получают (101) соответствующие аудиосигналы, представляющие речь пользователя, от множества аудиодатчиков, ианализируют (103, 105) соответствующие аудиосигналы для определения, какой, при наличии, из множества аудиодатчиков контактирует с пользователем устройства.2. Способ по п. 1, в котором этап анализа (103, 105) содержит этап, на котором анализируют спектральные свойства каждого из аудиосигналов.3. Способ по п. 1 или 2, в котором этап анализа (103, 105) содержит этап, на котором анализируют мощность соответствующих аудиосигналов выше пороговой частоты.4. Способ по п. 3, в котором определяют, что аудиодатчик контактирует с пользователем устройства, если мощность ...

УСТРОЙСТВО И СПОСОБ ДЛЯ ОПРЕДЕЛЕНИЯ ПОКАЗАТЕЛЯ ДЛЯ ВОСПРИНИМАЕМОГО УРОВНЯ РЕВЕРБЕРАЦИИ, АУДИОПРОЦЕССОР И СПОСОБ ДЛЯ ОБРАБОТКИ СИГНАЛА

... 1. Устройство для определения показателя воспринимаемого уровня реверберации в смешанном сигнале, состоящем из прямой компоненты (100) сигнала и реверберационной компоненты (102) сигнала, содержащее:процессор (104) модели громкости, содержащий каскад перцепционного фильтра для фильтрации «сухой» компоненты (100) сигнала, реверберационной компоненты (102) сигнала или смешанного сигнала, причем каскад перцепционного фильтра выполнен с возможностью моделирования механизма слухового восприятия объекта для получения фильтрованного прямого сигнала, фильтрованного реверберационного сигнала или фильтрованного смешанного сигнала;блок оценки громкости для оценки первого показателя громкости с использованием фильтрованного прямого сигнала и для оценки второго показателя громкости с использованием фильтрованного реверберационного сигнала или фильтрованного смешанного сигнала, где фильтрованный смешанный сигнал получен из суперпозиции прямой компоненты сигнала и реверберационной компоненты сигнала; ...

Verfahren und Vorrichtung zur Audiosignalklassifizierung

EICHVORRICHTUNG FUER MIKROPHONE

VERBESSERTES KÜNSTLICHES OHR UND OHRKANALSYSTEM UND VERFAHREN ZU SEINER HERSTELLUNG

Akustiksystem mit automatischer Umschaltung

Das automatische Schalten von Akustiksystemen und insbesondere von Hörgeräten in verschiedene Verarbeitungsprogramme soll verbessert werden. Hierzu ist vorgesehen, dass die Signalverarbeitungseinrichtung des Akustiksystems einen Hochfrequenzdetektor (HFD) zur Analyse eines Eingangssignals aufweist, dessen Ausgangssignal zur Verarbeitung eines oder mehrerer Eingangssignale durch die Signalverarbeitung nutzbar ist. Speziell kann hierdurch die Anwesenheit eines aktiven Mobiltelefons festgestellt werden, das elektromagnetische Strahlung im Hochfrequenzbereich abgibt. Dadurch kann mit hoher Sicherheit situationsgerecht in einen Telefonmodus geschaltet werden.

Schallfeld-Kompensationsvorrichtung und Schallfeld-Kompensationsverfahren

Die Schallfeld-Kompensationsvorrichtung weist einen Abschnitt auf, welcher ein Lautstärke regulierendes Testsignal erzeugt; einen Ansteuerabschnitt, welcher einen Lautsprecher ansteuert; ein Mikrophon, welches das Ausgangssignal vom Lautsprecher empfängt; und einen Steuerabschnitt, welcher ein Ausgangssignal des Mikrophons verarbeitet und der Betriebe von jeweiligen Abschnitten steuert. Das Tonlautstärke regulierende Testsignal ist ein Summensignal, welches für die Summe von Einzelfrequenz-Sinusschwingungssignalen repräsentativ ist, deren Frequenzen auf die Beziehung eines ganzzahligen Verhältnisses gesetzt sind. Der Steuerabschnitt bewirkt, dass der Ansteuerabschnitt den Lautsprecher unter Verwendung des Tonlautstärke regulierenden Testsignals ansteuert, um dadurch Signalpegel von Frequenzkomponenten der Sinusschwingungsformsignale von Ausgangssignalen des Mikrophons zu ermitteln. Gemäß einem Durchschnittswert der Signalpegel setzt der Steuerabschnitt eine Messtonlaustärke, wenn der Lautsprecher ...

X-System 2

Socket monitoring

Method and system for analysing audio tracks

Improvements in or relating to electro-acoustic devices

... 1,132,155. Testing receivers. STANDARD TELEPHONES & CABLES Ltd. 18 March, 1966, No. 11979/66. Heading H4J. A rocking armature telephone receiver is tested by applying a single-frequency audiofrequency signal to the energizing coils of the receiver, together with a biasing current to reduce the sensitivity of the receiver, and the resultant output of the receiver is used to determine the length of one or both gaps. In one arrangement, current varying about zero at approximately 5 c/s. with a maximum amplitude sufficient to bias the armature towards either of the polepieces by a distance equal to a predetermined minimum air gap length (e.g. 2À5 mils.) is applied to the energizing coils of the telephone receiver in parallel with a 1000 c/s. signal current. If the air gaps are equal and greater than the desired minimum the receiver sensitivity is reduced by equal amounts by both positive and negative bias current. If one air gap is less than the other the sensitivity is reduced at the extreme ...

Method and device for Measuring the Reference Equivalent or Articulation of Telephone Sets

... 1,175,740. Testing telephone transducers. TESLA NARODNI PODNIK. 5 May, 1967 [18 May, 1966], No. 20992/67. Heading H4J. Characteristics of the transducers of telephone sets are measured by apparatus comprising a signal generator 1 energizing at least two parallel connected filters 2, 3. Where two filters are used, one. may pass frequencies below 1000 c/s. and the other above 1000 c/s. Two band pass filters having frequency ranges of 200 to 1000 c/s. and 1000 to 4000 c/s. may alternatively be used. Alternatively there may be four parallel filters having bands of 200 to 450 c/s., 450 to 850 c/s., 850 to 1900 c/s. and 1900 to 4000 c/s. or bands of 200 to 850 c/s. 850 to 1600 c/s., 1600 to 2300 c/s. and 2300 to 4000 c/s. A switch 4 cyclically connects all the filters to a switch 5 in a period of one second. Contact a connects the signal to an amplifier 6 from where it is converted to sound in an artificial mouth 7. The microphone 8 of a telephone set 9 converts this sound to an electrical signal ...

Audio stream testing apparatus and a method thereof

Ultrasonic transudcer structures

Processing audio input signals

Processing audio input signals

Hydrophone calibration system

A chamber comprising a base and cap configured to contain a liquid under pressure, an aperture is provided in the cap through which a hydrophone (figure 1: 5) is exposed to the liquid, a reference sensor 10 is provided in the chamber and a transducer to apply an acoustic pressure to the liquid. The transducer may be a piston 8 and it may generate waves over a plurality of frequencies. A valve may be provided to decouple to reference sensor 10 or the piston 8 from the rest of the chamber, the valve may be a slide valve. The temperature of the chamber may be determined and a thermal bath may be coupled to the pressure chamber. The piston controller may be configured to measure the temperature difference between the thermal bath and the chamber and only actuate the piston 8 when the difference is below a predetermined limit. The liquid in the chamber may be degassed. The calibration may take place at acoustic frequencies as low as 0.001 Hz and at hydrostatic pressures of at least 5000 psi.

Audio device arrays in convertible electronic devices

Methods, apparatus and systems for audio playback

The present invention relates to methods, apparatus and systems for audio playback via a personal audio device following a biometric process. A personal audio device may be used to obtain ear model data for authenticating a user via an ear biometric authentication system. Owing to that successful authentication, the electronic device is informed of the person who is listening to audio playback from the device. Thus the device can implement one or more playback settings which are specific to that authorised user.

Audio circuitry

Single channel sampling for multiple channel vehicle audio correction

A hearing wellness monitoring system and method

A hearing wellness monitoring system which comprises at least one audio capture device, which is configured to provide sound data about an environment as a function of time, and a remote server. The audio capture device comprises a microphone and a communication interface which transmits the sound data from the audio capture device. The server receives the sound data and analyses the data to dynamically calculate a plurality of sound data parameters and calculate a hearing wellness parameter from the plurality of sound parameters. The hearing wellness parameters allow the assessment of a hearing wellness characteristic of the environment. Further inventive concepts include; obtaining hearing preference data from a user and assessing the suitability of an environment for the user based on the hearing preference data and a hearing wellness category of the environment, matching an environment to the user from a plurality of environments, and displaying information using a graphical user interface ...

PROCEDURE AND DEVICE FOR FORESEEING THE BEHAVIOR OF A TRANSDUCER

Method and system for measuring in-ear effective sound exposure under an earplug or without an earplug and for determining a wearer induced disturbance

Personalised calibration methods and devices adapted to identify acoustical corrections to be applied for in-ear dosimetry. The acoustical corrections allow to convert a measured acoustic pressure within the ear-canal of a user to an equivalent acoustic pressure at the ear-drum and/or in free-field. The methods and devices also allow to distinguish noises originating from the user from noises of the environment in the ear occluded by an earplug. Such a distinction is done using two microphones to simultaneously measure the acoustic pressure within the ear-canal and outside the ear-canal. The device may be used to determine a cumulative sound pressure level dosage for which a user being exposed to surrounding sounds over a predetermined period of time.

Method and apparatus for objective assessment of in-ear device acoustical performance

Unitary transducer control system

Identification system and method for recognizing any one of a number of different types of devices

Transfer device, electronic signature tool, detection device and interface detection system

Provided is an audio signal transfer device. The device comprises: a first interface, a second interface, an audio signal amplification circuit and an impedance circuit. The first interface comprises an audio signal receiving pin, a microphone pin, and a ground pin. The second interface comprises an audio signal output pin. The audio signal amplification circuit is used for amplifying the audio signals received by the audio signal receiving pin and outputting same to the audio signal output pin. The audio signals received by the audio signal receiving pin flow to the microphone pin or flow to the ground pin through an impedance equivalent part of the audio signal amplification circuit. The impedance circuit is coupled between the microphone pin and the ground pin. The audio signal transfer device according to the present invention can detect whether the wire sequence of a microphone pin and a ground pin of an audio plug of the transfer device is the same as the wire sequence of a detection ...

Speaker operation confirmation device

A speaker operation confirmation device that confirms operation of a right speaker attached on the right side of a vehicle and a left speaker attached on the left side of the vehicle, and comprises: a confirmation sound acquisition unit that obtains confirmation sounds to be generated by the right speaker and the left speaker; and a confirmation sound generation unit that causes the right speaker and the left speaker to generate the confirmation sound obtained by the confirmation sound acquisition unit. The confirmation sound has the volume of one out of the right speaker and the left speaker getting quieter over time and the volume of the other out of the right speaker and the left speaker getting louder.

NOISE POLLUTION MONITORING SYSTEM WITH DISPLAY PANEL AND VIDEO CAPTURE

Système de surveillance de pollution sonore comprenant : la détection par un audiomètre à distance d'un type et d'un niveau de bruit déterminé, l'affichage de celui-ci sur un panneau, la prise de photos de l'évènement déclencheur, et la transmission des données relative à l'évènement (heure, date, type de bruit, photo) à un serveur distant.

SOUND MANAGEMENT METHOD AND SYSTEM

A computer implemented method for managing a sound emitting device comprising: receiving data associated with operation of the sound emitting device at a predetermined location; processing said data to determine an operating characteristic of that device for that location; comparing the operating characteristic with a predetermined mathematical relationship to determine whether a difference exists; and identifying an input adjustment to correct the difference wherein the input adjustment optionally is within a predetermined range and optionally does not exceed a predetermined maximum increment; wherein the predetermined mathematical relationship is between an input variable and an output variable in respect of the sound emitting device.

INDUSTRIAL AUDIO NOISE MONITORING SYSTEM

A wireless field device (100) for use in monitoring acoustic noise includes an acoustic sensor (102) configured to sense acoustic noise. Processing circuitry (114) coupled to the acoustic sensor (102) is configured to identify a hazardous noise condition based upon the sensed acoustic noise and a personnel noise exposure standard. Output circuitry (106) provides a warning output in response to an identified noise condition. A system (700) is also provided which uses one or more acoustic sensors implemented in wireless field mounted monitors.

AUDIO SIGNAL VERIFICATION FOR VIDEO/AUDIO PRODUCTION EQUIPMENT

Audio signal verification methods and apparatus for video/audio production equipment are disclosed. An audio signal and a video signal are input into production equipment. The audio signal has one or more known frequencies, and the video signal has a known video frame to which the audio signal is to be synchronized by the equipment. Based on the known frequency or frequencies and a frequency or frequencies of an output audio signal that is output by the equipment with the video frame, a determination can be made as to whether the output audio signal corresponds to the input audio signal. Respective values could be assigned to a number of frequencies, to enable values to be encoded in the input audio signal. The correspondence determination could then be made based on whether the values encoded by frequencies in the output audio signal match the values encoded in the input audio signal.

LOUDSPEAKER LINE ARRAY CONFIGURATIONS AND RELATED SOUND PROCESSING

A sound reproduction system includes one or more arrays of drive units, coupled with sound processing allowing improved steerability, directional control, width control, and/or beam overlay. A speaker column may comprise two arrays facing one another, the drive units being perpendicular to the speaker unit front, with the acoustic output of each line array being compressed, turned and output from an elongate slot. The drive units may be staggered with respect to those in the opposing array. The arrays may be arc-shaped else straight. Selective delays to simulate an arced pattern. Differential delays applied to the drive units in a sub-array allow beam steering. Additional drive units, such as high frequency drivers, may be added along the length of the elongate slot or elsewhere to increase the dynamic frequency range of the speaker system. A collocated sound processor and amplifier output stage may be integrated with the speaker unit.

LOUDSPEAKER LINE ARRAY CONFIGURATIONS AND RELATED SOUND PROCESSING

A sound reproduction system includes one or more arrays of drive units, coupled with sound processing allowing improved steerability, directional control, width control, and/or beam overlay. A speaker column may comprise two arrays facing one another, the drive units being perpendicular to the speaker unit front, with the acoustic output of each line array being compressed, turned and output from an elongate slot. The drive units may be staggered with respect to those in the opposing array. The arrays may be arc-shaped else straight. Selective delays to simulate an arced pattern. Differential delays applied to the drive units in a sub-array allow beam steering. Additional drive units, such as high frequency drivers, may be added along the length of the elongate slot or elsewhere to increase the dynamic frequency range of the speaker system. A collocated sound processor and amplifier output stage may be integrated with the speaker unit.

APPARATUS AND METHOD FOR DETERMINING A MEASURE FOR A PERCEIVED LEVEL OF REVERBERATION, AUDIO PROCESSOR AND METHOD FOR PROCESSING A SIGNAL

An apparatus for determining a measure for a perceived level of reverberation in a mix signal consisting of a direct signal component (100) and a reverberation signal component (102), comprises a loudness model processor (104) comprising a perceptual filter stage for filtering the dry signal component (100) the reverberation signal component (102) or the mix signal, wherein the perceptual filter stage is configured for modeling an auditory perception mechanism of an entity to obtain a filtered direct signal, a filtered reverberation signal or a filtered mix signal. The apparatus furthermore comprises a loudness estimator for estimating a first loudness measure using the filtered direct signal and for estimating a second loudness measure using the filtered reverberation signal or the filtered mix signal, where the filtered mix signal is derived from a superposition of the direct signal component and the reverberation signal component. The apparatus furthermore comprises a combiner (110) ...

METHODS AND SYSTEMS FOR IMPROVED ACOUSTIC ENVIRONMENT CHARACTERIZATION

In embodiments of the present invention improved capabilities are described for determining a multi-dimensional sound signature for a location within a hypothetical space, comparing the multi-dimensional sound signature to a known multi-dimensional sound signature, and modifying the hypothetical space such that the similarity between the multi-dimensional sound signature for the location within the modified hypothetical space and the known multi-dimensional sound signature is increased.

AUDIO POWER MANAGEMENT SYSTEM

An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

SYSTEM AND METHOD FOR PROTECTING TRANSMISSIONS OF WIRELESS MICROPHONES OPERATING IN TELEVISION BAND WHITE SPACE

A wireless microphone system broadcasts a pilot tone at the designated ATSC pilot position in the TV Band channel being used by the wireless microphone system. The pilot tone is a readily detectable waveform transmitted in the ATSC pilot position. The pilot tone can be generated by any one of: a standalone pilot tone generator; a pilot tone generator incorporated into a wireless microphone receiver; or, a pilot tone generator incorporated into a wireless microphone.

VIRTUAL AUDIO MAP

A system and method for mapping acoustic activities is disclosed. One method comprises receiving acoustic measurements recorded for a predetermined interval at a plurality of telephonic computing devices that are located at a determined location. A map of the acoustic measurements is created based on a location of each of the plurality of telephonic computing devices at which the acoustic amplitude measurement is recorded. An amplitude and/or frequency of the acoustic measurements can be displayed on the map relative to the location of the telephonic computing device at the selected location to form an acoustic map.

SYSTEM AND METHOD FOR HEARING AID APPRAISAL AND SELECTION

The selection of hearing aids can be based on many factors which are beyond a simple compensation for the hearing impairment of the user. The process of selecting hearing aids can be time consuming and resource intensive. By rapidly and effectively providing the necessary context to hearing-impaired individuals to evaluate the benefit provided by various hearing aids, without the traditional time consuming fitting trial, the purchase of a device can be expedited and provide the necessary correction to negate the effects of auditory deprivation. A system and method for enabling improved hearing aid appraisal and selection processes is provided.

Kastenförmiger abgeschirmter Messraum zur Messung von elektromagnetischen oder akustischen Wellen

Verfahren zum Herstellen eines elektro-akustischen Wandlers

Dispositif détecteur à seuil des vibrations d'un haut-parleur

Coupler for measuring a hearing aid and method for measuring a RECD.

Die Erfindung betrifft einen Kuppler (20) zum Ausmessen eines Hörgerätes (1). Der Kuppler (20) weist ein Mikrofon (11) auf, mit einer Membran (12), welche ein Frontvolumen (13) von einem Backvolumen (14) trennt. Eine Frontplatte (17) weist einen Stutzen (18) auf. Der Stutzen (18) ist derart gestaltet, dass das Hörgerät (1) nach Entfernen eines Cerumenschutzes auf den Stutzen (18) aufgesteckt werden kann. Optional weist der Stutzen (18) ein Sieb (19) auf, welches die akustischen Eigenschaften des entfernten Cerumenschutzes nachbildet. Die Erfindung betrifft ferner ein Verfahren zum Messen einer RECD für ein Hörgerät (1) an einem Ohr einer Person.

Information signal representing apparatus

Method for estimating the battery life of battery-powered audio systems by means of an amplitude modulated broadband audio signal

Method and apparatus for audio signal classification

System and method for audio tests

AUDIO SIGNAL PROCESSING APPARATUS

Dispositif de traitement (1) d'un signal audio initial (4), comprenant : • un bloc détecteur d'aigus (2) apte à analyser le signal audio initial (4) pour déterminer un taux d'aigu (%), • un bloc filtre (3) apte à atténuer le signal audio initial (4) dans les aigus, si le taux d'aigu (%) est inférieur à un seuil maximal (Smax).

음질평가장치 및 그 방법

... 본 발명의 실시예에 따른 음질평가장치는 악기에 의해서 연주되는 음질을 평가하는 장치로서, 음향이 유입되는 음향 수신부, 상기 음향 수신부에 유입되는 시간영역에서의 음향정보를 주파수영역의 주파수분포정보로 변환하는 정보변환부, 상기 주파수분포정보를 이용하여 기음과 배음을 추출하는 주파수 추출부 및 상기 기음과 상기 배음을 이용하여 음질을 평가하는 음질 평가부를 포함한다.

이형부, 인공 두부 및 이들을 이용한 측정 시스템 및 측정 방법

... 진동 소자(102)를 구비하고, 진동 전달에 의해 소리를 들려주는 음향 장치(100)를 평가하기 위한 측정 시스템으로서, 인체의 귀를 본뜬 귀 모형(51) 및 귀 모형(51)에 접하는 인공 측두골부(57)를 구비한 이형부(50)와, 인공 측두골부(57)에 배치된 진동 검출 소자(56)를 구비한다.

AUDIO JACK INSERTION OR REMOVAL FAULT DETECTION

Discussed are circuits and methods for determining whether moisture exists in an audio socket by displaying impedance between a detection pin and a first signal pin of an audio socket by using first and second comparators. When moisture exists in the audio socket, communication between an audio processing unit and the audio socket can be disabled. COPYRIGHT KIPO 2016 ...

Sound-based brake wear detection for a vehicle braking system

ACOUSTIC IN-EAR DETECTION FOR EARPIECE

SYSTEM FOR HEADPHONE EQUALIZATION

APPARATUS FOR EVALUATING SOUND QUALITY AND METHOD THEREOF

According to an embodiment of the present invention, an apparatus for evaluating a sound quality evaluates a sound quality produced by an instrument. The apparatus for evaluating a sound quality includes: a sound reception unit into which a sound is introduced; an information conversion unit which converts sound quality information in a time domain of being introduced to the sound reception unit into frequency distribution information of a frequency region; a frequency extraction unit which extracts a ground note and a harmonic tone by using the frequency distribution information; and a sound quality evaluation unit which evaluates a sound quality by using the ground note and the harmonic tone. COPYRIGHT KIPO 2016 (100) Sound reception unit (200) Information conversion unit (300) Frequency extraction unit (400) Sound quality evaluation unit ...

INFORMATION SIGNAL REPRODUCING DEVICE

PURPOSE: To provide an information signal reproducing device in which optimum reproducing environment is formed without being adversely affected by the difference in the signal format of audio data to be reproduced or the like and the difference in a recording medium and by which an easily usable reproducing system can be structured. CONSTITUTION: The type of an optical disk 100 and the signal format information of audio data to be read and reproduced from the disk 100 are detected by an RF signal processing circuit 106 and a video/audio decoding section 107. On the basis of the detected information, a system control section 114 presumes whether the audio data to be reproduced are musical sound or video sound. On the basis of the estimated result, the section 114 and an interface control section 117 control an audio signal processing section 109, an audio volume 111 and a speaker amplifier 112 so as to conduct signal processing in accordance with the audio data to be reproduced. © KIPO ...

sistema e método para teste acústico

Method and system for audio calibration of an audio device

A method for audio calibration of an audio device is provided. The method comprises: registering, by a microphone (108), a representation of an audio signal (114); processing, by a data processing unit, the registered representation of the audio signal (114) into feedback image data; displaying on a display (112), the feedback image data; capturing, by a camera (106), the feedback image data displayed on the display (112); altering a setting affecting audio of the audio device (102) based on the feedback image data captured by the camera (106). A system for audio calibration of an audio device is further provided.

Combo-jack detecting circuit and operating method thereof

LISTENING DEVICE

It is an object to provide a listening device that can surely measure living body information even during the activity of a living body, stabilizes measured results of the living body information, and obtains a good SN ratio. A listening device is provided with an electro-acoustic transducer unit that is received in a speaker housing (3) and an ear pad (9) that is set at the speaker housing. When a main body of the listening device is set, a self-light emitting sensor (20) for optically measuring living body information of an examinee is provided at a portion of the ear pad (9) corresponding to the auricle of the examinee.

FULL FUNCTION TEST FOR IN SITU TEST OF SENSORS AND AMPLIFIERS

The present invention provides a method and an apparatus for in situ test of transducers comprising sensing elements and associated conditioning preamplifiers. The invention makes it possible to evaluate the characteristics of the complete transducer by means of higher integration of the transducer circuitry. Tests can be performed from a remote central location without additional wiring and while the transducer is in operating environment., Testing is performed by superposing test signals and test sequence control signals on the wiring for the transducer output signal, hereby offering flexibility without sacrificing simplicity. Test signalling is by additional circuitry in the transducer interpreted and routed to the input of the conditioning preamplifier based on signalling from the remote test generator, and the signals engendered from the test signals can be analyzed from a remote analyzing system for complete qualifications of the transducer under test.

BUILT-IN AMBIENT SOUND SYSTEM FOR MANAGING AND REPRODUCING AUDIO IN A COMPRESSED DIGITAL FORMAT

The invention relates to a built-in ambient sound system for managing and reproducing audio in a compressed digital format, characterized in that: it uses the compressed digital audio format internally, with no need to connect external devices such as mp3 players, iPods, flash drives, etc; it includes a central module and peripheral modules connected to each other by means of a wired or wireless data communications network; it includes a means for connecting the system to the Internet and providing Internet access for all the devices in the system independently; it includes non-volatile memories for storing audio in a compressed digital format; it includes a means for wirelessly connecting external devices such as iPods, mobile phones or others; it is connected to a domotic system for controlling the domotic parameters from any peripheral module; and it is connected to a video intercom for displaying the image on the video intercom, answering calls and controlling access, all from any peripheral ...

A DEVICE COMPRISING A PLURALITY OF AUDIO SENSORS AND A METHOD OF OPERATING THE SAME

There is provided a method of operating a device, the device comprising a plurality of audio sensors and being configured such that when a first audio sensor of the plurality of audio sensors is in contact with a user of the device, a second audio sensor of the plurality of audio sensors is in contact with the air, the method comprising obtaining respective audio signals representing the speech of a user from the plurality of audio sensors; and analyzing the respective audio signals to determine which, if any of the plurality of audio sensors is in contact with the user of the device.

METHOD AND SYSTEM

A method of warning and/or reduction of risk of ear damage of a hearer by excessive exposure to relatively high volume and/or frequency of sound, wherein a non-audio input representative of desired sound and/or ambient sound is employed to warn of said risk and/or said risk is automatically reduced.

Earpiece for acoustical source and load modeling

A wearable device configured to be modeled as both a part of the source and a part of the load in an electro-acoustical source-and-load model of a system where a sound source is coupled to the ear canal. Such a wearable device may be used to determine the acoustic energy density at a point of measurement in or near the ear of a user.

System and method for reducing power consumption in an audio system by disabling filter elements based on signal level

An audio system includes a digital audio interface, a first filter stage, a second filter stage, a controller, and a first path selector. The controller is communicatively coupled to the digital audio interface and configured to determine whether an audio signal received at the digital audio interface is below a first signal threshold. The controller selects one of a first signal path that passes through the first and second filter stages and a second signal path that bypasses the first filter stage based upon the determination. The first path selector is communicatively coupled to the controller and at least one of the first filter stage and the second filter stage and is configured to generate the selected one of the first and second signal paths.

Information signal reproducing apparatus

An information signal reproducing apparatus which is capable of forming a suitable reproducing environment and building an easy-handling reproducing system regardless of differences such as signal format or recorded media of audio data to be reproduced is provided, in which: a medium type of an optical disc and signal format information of audio data to be read from the optical disc and to be reproduced are detected by an RF signal processing circuit and a video/audio decoding unit so that a system control unit estimates whether the audio data to be reproduced is a music sound, a movie sound or another type of audio data in accordance with the detected information, and, in accordance with this estimation result, the system control unit and an interface control unit control an audio signal processing unit, an audio volume unit and a speaker amplifier unit so as to carry out a signal processing suitable for the audio data to be reproduced.

Test Device And Test Method For Active Noise Reduction Headphone

The present invention discloses a test device and test method for the noise reduction headphone. The test device comprises: an enclosed cavity, a noise source, a test panel, a measuring microphone and a measure comparison module connected with the measuring microphone. The sound emitted from the noise source is sealed within the enclosed cavity. The test panel can cooperate with the noise reduction headphone to form a coupling cavity in the test. The test panel has a sound guiding hole in the common part with the enclosed cavity for transmitting the sound of the noise source into the interior of the coupling cavity. The test panel also has a mounting hole, and the measuring microphone is mounted on the mourning hole towards the direction of the coupling cavity. The measuring microphone records noise signals before and after the noise reduction function of the noise reduction headphone is activated. The measure comparison module receives the signals recorded these two times by the measuring ...

Apparatus and method for testing sound input and output of sound card

An apparatus and a method for testing a sound card are applicable for detecting whether a sound leakage problem occurs to a sounding interface of the sound card. The testing apparatus includes a switching circuit and a switching unit. The switching circuit is electrically connected to the sounding interface and a sound-receiving interface of the sound card respectively, and has a plurality of transfer paths. The switching circuit is used to receive a first and a second sound channel, and to connect the first and the second sound channel to corresponding transfer paths, so as to transmit the audio signal back to the sound-receiving interface. The switching unit is electrically connected to the sounding interface and the switching circuit, for receiving the first and the second sound channel. The switching unit alters the transfer paths in the switching circuit according to a first and/or a second switching audio signal.

System and method for mapping and displaying audio source locations

The present invention relates generally to a method and system for defining a reference sound position and producing an indicia proximate thereto in relation to one or more sound characteristics. The present invention, in one or more embodiments, provides for displaying a holographic image at a reference sound location which is determined in relation to the identification of one or more target sounds being associated with one or more identified sound characteristics. In other embodiments, the present invention provides for an indicia to appear to be interactive with a reference sound location and may be used in a variety of environments including but not limited to rock theatrics, homeland security and residential security.

Method and apparatus to evaluate audio equipment for dynamic distortions and or differential phase and or frequency modulation effects

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.

Method of determining operation mode of hearing device and hearing device

A method of determining an operation mode of a hearing device and the hearing device may include detecting an input signal of the hearing device, determining the operation mode of the hearing device by determining whether the input signal is a designated signal related to a programming mode of the hearing device, and controlling the hearing device according to the operation mode.

Vibration source estimation device, vibration source estimation method, and vibration source estimation program

A vibration source estimation device, method and computer-readable medium are provided. The vibration source estimation device comprises a processor configured to execute instructions to receive data samples of a plurality of vibration signals generated at a vibration position, the data samples being associated with a plurality of frames. The processor is also configured to execute instructions to calculate a cross correlation function for each frame based on the data samples, calculate a weight for each frame based on a signal-to-noise ratio calculated from the cross correlation function, multiply the cross correlation function for each frame by the weight, calculate a sum of weighted cross correlation functions for respective frames as a weighted-cross-correlation function, and estimate the vibration position based on the weighted-cross-correlation function.

Headset system comprising a noise dosimeter

A noise dosimeter (xOO) for monitoring the exposure to noise of a user wearing a headset is described. The noise dosimeter includes a housing to be worn external to the ear of the user, an earpiece (xO1) for placement in or near an ear canal of the user, a microphone transducer, and a signal transmission means (xO3) for transmission of a signal from the earpiece (xO1) to the housing. The earpiece (xO1) comprises sound collection means (xO4) for collecting sound emitted from a speaker of the headset, and the housing comprises signal processing circuitry for processing and accumulating signals from the microphone transducer in order to evaluate a user's exposure to noise when using the headset. The noise dosimeter can be integrated in a headset.

AUDIO INTERFACE CIRCUITS AND METHODS

In one embodiment, an electronic device includes an audio connector port comprising a ground terminal and one or more audio output terminals, a first audio amplifier coupled to one of the audio output terminals, and a multiplexer having an output terminal coupled to the input terminal of the first audio amplifier. Sense circuits inside the electronic device may be alternately coupled through the multiplexer and first audio amplifier so that, in a first mode of operation, the multiplexer couples the audio signal to the first audio output terminal, and in a second mode of operation, the multiplexer couples an analog voltage corresponding to an internally sensed value to the first audio output terminal. Use of the audio connector and audio circuitry to access internal electrical parameters may facilitate testing and analysis of internal systems.

EAR INPUT SOUND PRESSURE LEVEL MONITORING SYSTEM

Systems and methods for monitoring a sound pressure level dose at an ear are provided. A system includes an audio transducer which outputs a sound signal is placed within an ear to receive sound at the ear. A sound level threshold detector determines whether a sound pressure level of the sound signal is at a minimum level and outputs a sound pressure level signal corresponding to the sound pressure level when the sound pressure level is not at the minimum level. A time period is calculated during which the sound pressure level is not at the minimum threshold level. A listening fatigue calculator determines whether a cumulative effect of exposure to the sound signal at the ear over the time period will cause harm to the ear. At least a portion of the system is disposed in situ at the ear.

WIRELESS EARPHONE SYNCHRONIZATION DETECTION METHOD, APPARATUS, WIRELESS EARPHONES AND STORAGE MEDIUM

A method and a device for performing synchronization detection on wireless earphones, wireless earphones and a computer readable storage medium are provided. In the method, a synchronization state between a left earphone and a right earphone of the wireless earphones is acquired when the wireless earphones trigger a target action, where the target action includes any one of: entering a box, leaving a box, entering an ear, leaving an ear and connecting to a terminal; a countdown with a predetermined time duration is started in a case that the synchronization state is a non-synchronous state; and the synchronization state between the left earphone and the right earphone of the wireless earphones is acquired when the countdown ends.

In-ear detection utilizing earbud feedback microphone

A method for in-ear detection, the method may include transmitting test signals, by a speaker of an earbud, during a test period, and while the earbud is operating at a first operational mode, wherein the test signals comprise at least one first test signal within a first frequency range, at least one second test signal within a second frequency range, and at one third test signal within a third frequency range; wherein the first frequency range, the second frequency range and the third frequency range differ from each other and are within a human auditory range; generating, by a feedback microphone of the earbud, sensed information that is indicative of audio signals sensed by the feedback microphone as a result of the transmitting of the test signals; and determining whether the earbud is located within an ear of a person, wherein the determining is based on the sensed information and a reference out of ear spectrum.

Group Coordinator Selection

A first playback is configured to (i) determine that it is connected to a local data network using a first networking protocol, (ii) receive a command to join a second playback device in a group of playback devices that are configured for synchronous playback of audio content from an audio source, (iii) determine that the second playback device is connected to the local data network using a second networking protocol different from the first networking protocol, (iv) based on determining that the second playback device is connected to the local data network using a second networking protocol different from the first networking protocol, configure the first playback device as a group coordinator of the group of playback devices, and (v) while rendering the audio content, (a) receive the audio content from the audio source and (b) forward at least a portion of the received audio content to the second playback device.

Method and apparatus for objective assessment of in-ear device acoustical performance

A method and apparatus for objectively assessing acoustical performance of an in-ear device having a passageway extending there through use a dual microphone probe that removably engages the passageway. The acoustical performance of the in-ear device is performed with the in-ear device inserted into the ear canal of the user and a reference sound source. A clip holding the probe in an acoustic near field of the sound source permits real time calibration thereof. The method and apparatus allow on-site and in-situ measurement of a predicted personal attenuation rating of the device, a subject-fit re-insertion test, an acoustic seal test, a rating test, a stability and reliability test, as well as a protection test of the device with an assessment of a filtered predicted exposure level at the ear for a specific noise exposure level. The apparatus may be simply housed along with the sound source for in-field evaluation tests.

METHODS AND APPARATUS TO MONITOR A MEDIA PRESENTATION

Methods and apparatus for gathering research data indicating exposure to media are disclosed. An example apparatus includes a first wireless receiver to access a first modulated audio signal via a wireless data connection from a media device using a wireless communication protocol, the first modulated audio signal associated with the media, the first wireless receiver to output a baseband audio signal corresponding to the first modulated audio signal. The example apparatus includes a first wireless transmitter to modulate the baseband audio signal to form a second modulated audio signal, the first wireless transmitter to transmit the second modulated audio signal to wireless headphones using the wireless communication protocol. The example apparatus includes a monitoring device to receive the baseband audio signal, and to generate research data identifying the media, the media device including a second wireless transmitter to transmit the first modulated audio signal to the first wireless ...

AUDIO SIGNAL PROCESSING METHOD AND SYSTEM

CHARACTERISTIC MEASURING INSTRUMENT FOR PICKUP CARTRIDGE

PURPOSE: To measure characteristics of a pickup cartridge itself with accuracy by vibrating the stylus of the pickup cartridge in various directions. CONSTITUTION: A cutter head is used as the exciter and while the stylus of the pickup cartridge is set in hole 1 at the tip of cantilever 2, vibration is applied via driving shafts 3 and 4 in right and left 45-degree directions to measure frequency characteristics and physical characteristics such as strain and crosstalk between right and left channels. COPYRIGHT: (C)1981,JPO&Japio ...

Digital Data Transfer Via Audio Signal Conductors

An electro-acoustic transducer or a device into which an electro-acoustic transducer is incorporated has at least two audio signal conductors through which the electro-acoustic transducer or device may be coupled to another device to convey analog audio signals representing sounds at a time when the other device places a relatively low DC bias voltage across the at least two audio signal conductors, and to be operated as a digital serial bus to exchange pieces of digitally-encoded data concerning the electro-acoustic transducer at a time when the other device places a relatively high DC bias voltage across the at least two audio signal conductors.

Personalized adjustment of an audio device

Described herein are apparatuses, systems and methods that facilitate user adjustment of an audio effect of an audio device to match the hearing sensitivity of the user. The user can tune the audio device with a minimum perceptible level unique to the user. The audio device can adjust the audio effect in accordance with the minimum perceptible level. For example, a loudness level can adjust automatically to ensure that the user maintains a perceptible loudness, adjusting according to environmental noise and according to the minimum perceptible level. Also described herein are apparatuses, systems and methods related to an audio device equipped with embedded audio sensors that can maximize a voice quality while minimizing the effects of noise.

Dust protection apparatus for flat loudspeakers

A method for fabricating flat loudspeakers comprising manufacturing a flat loudspeaker including at least one microspeaker array, having first and second main surfaces; and covering at least one of the main surfaces of the loudspeaker with a cover member including an airtight sound-pressure wave transparent thin polymer film.

System for extraction of reverberant content of an audio signal

A reverberant characteristic of an acoustic space is superimposed on an audio signal that is received by an apparatus. The apparatus decomposes the audio signal into an estimated original dry signal component and an estimated reverberant characteristic of the acoustic space. Estimation of the original dry signal component and the reverberant characteristic of the acoustic space is based on determination of an estimated impulse response of the acoustic space from the received audio signal. Once the audio signal is decomposed, the estimated original dry signal component and the estimated reverberant characteristic of the acoustic space may be independently modified by the apparatus. The modified or unmodified estimated original dry signal component and estimated reverberant characteristic of the acoustic space may be combined by the apparatus to produce one or more adjusted frequency spectra.

Apparatus, method, and program product for presenting moving image with sound

According to one embodiment, an apparatus for presenting a moving image with sound includes an input unit, a setting unit, a main beam former unit, and an output control unit. The input unit inputs data on a moving image with sound including a moving image and a plurality of channels of sounds. The setting unit sets an arrival time difference according to a user operation, the arrival time difference being a difference in time between a plurality of channels of sounds coming from a desired direction. The main beam former unit generates a directional sound in which a sound in a direction having the arrival time difference set by the setting unit is enhanced, from the plurality of channels of sounds included in the data on the moving image with sound. The output control unit outputs the directional sound along with the moving image.

Transitioning multiple microphones from a first mode to a second mode

An apparatus includes multiple microphones and a controller. The controller is coupled to receive a signal from each of the multiple microphones. The controller is configured to control a transition of the multiple microphones from an active mode to a dormant mode. When the multiple microphones are in the active mode, the controller is configured to perform signal processing responsive to signals received from at least two of the multiple microphones. When the multiple microphones are in the dormant mode, the controller is configured to select a microphone of the multiple microphones and to perform signal processing corresponding to the selected microphone while suspending signal processing corresponding to unselected microphones.

Integrated speaker and home theater sound system having the same

An integrated speaker and a home theater sound system having the same are provided. The integrated speaker includes: a first speaker that is configured to stand on a floor; and a second speaker mounted to an upper end of the first speaker, wherein the second speaker is a height speaker.

Audio signal quality measurement in mobile device

Technologies are generally described for a method for measuring a quality of an audio signal in a mobile device. In some examples, the mobile device includes a receiving unit configured to receive an audio signal transmitted from another device; an audio quality measuring unit configured to measure a quality of the received audio signal; and a transmission unit configured to transmit the measured quality of the audio signal to the another device.

Transducer with Integrated Sensor

An electroacoustic transducer includes a housing and a moving portion adapted to move relative to the housing. The moving portion has a first surface and includes a first electrode. A second electrode on a first surface of the housing is located proximate to a first region of the first electrode, and a third electrode on the first surface of the housing is located proximate to a second region of the first electrode. A capacitance between the first and second electrodes and a capacitance between the first and third electrodes vary similarly with displacement of the moving portion relative to the housing. An impedance buffer is coupled to the second electrode and the third electrode.

Multiple superimposed audio frequency test system and sound chamber with attenuated echo properties

A composite sound dampening structure includes a first base layer of sound dampening material extending around and against an inside surface of a container and a second wedge layer of sound dampening material attached to an inside surface of the first base layer. The composite sound dampening structure provides improved acoustic dampening in relative small sound chambers. An audio test system generates a composite audio signal of multiple different audio signals that are combined together using linear superposition. The composite audio signal allows a device to be simultaneously tested with multiple different audio frequencies.

Audio device and method for audio outputting

An audio device includes a headphone jack to receive a headphone plug, an audio output unit connected to the headphone jack, a detecting unit, and a control unit. The detecting unit detects insertion of a headphone plug into the headphone jack when playing an audio file. The control unit directly decreases an output power of the output unit to a first preset value within a preset period in response to the detected insertion of a headphone plug into the headphone jack, and gradually increases the output power of the output unit to a second preset value after the preset period is elapsed. A related method is also provided.

Hearing aid device with a directional microphone system and method for operating a hearing aid device having a directional microphone system

A hearing aid device has a directional microphone system with a first microphone outputting a first microphone signal and a second microphone outputting a second microphone signal. A delay unit generates a directivity by delaying the second microphone signal or a fourth microphone signal derived therefrom by an internal time delay and associating it with the first microphone signal or a third microphone signal derived therefrom for generating a directional microphone signal. A cross-correlation analysis unit receives the first or the third microphone signal and the second or the fourth microphone signal and determines a value of a cross correlation of the two microphone signals. A control unit adjusting the time delay depending on the value of the cross correlation of the two microphone signals. A classifier determines audio conditions in which the hearing aid device is currently situated, and the time delay is adjusted depending on the audio conditions.

Apparatus and methods for testing of acoustic devices and systems

Methods and devices are disclosed for testing an acoustic probe having transducing elements for converting between acoustic and electrical signals. An electrical signal is generated at a frequency with a testing device capable of generating electrical signals over a range of frequencies. The electrical signal is transmitted to at least some of the transducing elements to measure a complex impedance and thereby evaluate a performance of the transducing elements.

Network system for audio equipment, method and computer program

Wireless microphone systems are used, for example, in theater, stage performances, television, broadcast radio or in concerts to amplify voices or instruments for a large audience. Due to the plurality of actors or musicians, a corresponding number of audio channels is required to transmit the audio information. 20 to 50 audio channels can thus be required in a parallel mode even in common applications. The invention relates to a network system ( 1 ) for audio equipment ( 4 ) having a plurality of base stations ( 3 ) and having a plurality of microphone modules ( 4 ) wherein the microphone modules ( 4 ) are and/or can be allocated to the base stations ( 3 ) via configuration data and wherein each microphone module ( 4 ) of a microphone-module-base-station allocation is designed to transmit audio information on a transmitting frequency F1, F2, F3, F4, FX, FY to the allocated base station ( 3 ) of the same microphone-module-base-station allocation, wherein the base stations ( 3 ) are networked to one another such that the configuration data are interchangeable, having a control device ( 2 ), wherein the control device ( 2 ) is designed in respect of programming technology and/or circuit technology to automatically adapt the configuration data of the base stations ( 3 ) and/or microphone modules ( 4 ) and/or microphone-module-base-station allocations. The control device ( 2 ) is designed to adapt the configuration data in consideration of measured and/or calculated interactions of the microphone-module-base-station allocations and/or in consideration of the intended use of the respective microphone module ( 4 ).

Orientation-Responsive Acoustic Array Control

An audio device incorporates a plurality of acoustic drivers and employs them to form either a first acoustic interference array generating destructive interference in a first direction from the plurality of acoustic drivers or a second acoustic interference array generating destructive interference in a second direction from the plurality of acoustic drivers in response to the orientation of the casing of the audio device relative to the direction of the force of gravity.

Audio testing system and method

An audio testing system is configured for receiving an audio signal from the an audio emitting device. The system samples the audio signal and obtains sampling points from the audio signal for determining if the audio signal has been distorted. A related method is also disclosed.

Fully supervised self testing alarm notification apparatus

A method of testing audio emergency notification apparatuses which are provide to alert occupants of a building that an emergency condition which requires action is occurring, includes activating a control panel manually or automatically to test an audio apparatus, instructing the control panel to test the audio apparatuses either one by one, sequentially or simultaneously and generating a test signal for testing the apparatuses which is sent via closed circuit or radio frequency to each audio apparatus. The test signal causes each apparatus to generate an audio signal, the level or frequency of which from each apparatus is measured with a calibrated microphone and is sent to a selected destination.

Acoustic Localization of a Speaker

A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

Multi-channel broadcasting system using digital matrix amplifier and method of controlling the same

A multi-channel broadcasting system using a digital matrix amplifier performing the setting of respective equipment is provided to control the model equipment through a digital master controller. An 8-channel audio input unit 66 is electrically connected to a digital audio mixer. Sound source equipment connection terminals are electrically connected to the sound equipment. A data converter unit 46 converts each data and a contact point signal. An audio matrix IC 60 outputs an input sound source signal to an arbitrary channel. A pair of audio output terminals transmits the output signal of the audio matrix IC.

Devices with enhanced audio

A system for enhancing audio including a computer and an output device. The computer includes a sensor configured to determine a user location relative to the computer. The sensor is also configured to gather environment data corresponding to an environment of the computer. The computer also includes a processor in communication with the sensor and configured to process the user location and the environment data and adjust at least one of an audio output or a video output. The output device is in communication with the processor and is configured to output at least one of the audio output or the video output.

Methods and apparatuses for echo cancelation with beamforming microphone arrays

Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of microphones oriented to cover a corresponding plurality of direction vectors and to develop a corresponding plurality of microphone signals. A processor is operably coupled to the plurality of microphones. The processor is configured to perform a beamforming operation to combine the plurality of microphone signals to a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals. The processor is also configured perform an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals and select one of the plurality of combined echo-canceled signals for transmission.

Headset plug universal auto switcher

One embodiment of the invention is directed to a switchable plug comprising a first incoming line configured to carry a signal from a source device, a second incoming line configured to carry a signal from the source device, a Mic Bias output line, a Ground output line, and a switch control. The switch control is operably connected with the first incoming line, the second incoming line, the Mic Bias output line and the Ground output line. The switch control is configured to detect a type of the signal carried on the first incoming line and a type of the signal carried on the second incoming line, from the source device. The switch control is further configured to automatically route a Mic Bias type signal to the Mic Bias output line and a Ground type signal to the Ground output line, without user intervention.

Optimized Volume Adjustment

A method for adjusting the sound volume of media clips using volume adjuster lines is provided. The volume adjuster lines are individually set for each clip based on the intrinsic, or absolute, volume values of the clip. In some embodiments, the volume adjuster lines are set for each clip based on the peak value or a calculated loudness equivalent of the clip. A user can move the volume adjuster line to set the absolute sound level of a clip. The volume adjuster lines can be hidden in some embodiments. In these embodiments, dragging on any portion of a clip is treated as dragging on the volume adjuster line. Some embodiments provide a deformable volume adjuster line, or curve. In these embodiments, a single audio clip can have several different volume adjuster lines for different sections of the clip where the volume adjuster line for each section is individually adjustable.

Mobile device for multi-channel sound collection and output using common connector, and driving method thereof

A mobile device for providing multi-channel sound collection and output using a common connector, and a driving method thereof are provided. A mobile device having a sound compressor and a sound decompressor includes a connector slidably receiving a sound collector plug or a sound output plug, a plug identification unit for, if one of the sound collector plug and the sound output plug is coupled to the connector, identifying if the plug coupled to the connector is the sound collector plug or is the sound output plug, a switching unit for electrically coupling the plug coupled to the connector, to the sound compressor, or electrically coupling the plug coupled to the connector, to the sound decompressor, and a controller for controlling switching of the switching unit according to the identification outcome.

SOUND PRESSURE ASSESSMENT SYSTEM, AND METHOD AND PROGRAM THEREOF

A sound pressure assessment system determines sounds to be presented by referring to a database retaining data of sounds, each of which is a pure tone. The system presents the determined sound as a first sound, and in a predetermined time after presenting the first sound, presents as a second sound a sound at least having the same frequency and the same sound pressure level as those of the'first sound to the user. The system extracts an amount of change from an N1 component in response to the first sound to that in response to the second sound. The N1 components are negative components of an user's event-related potential based on points of presenting the first and second sounds as respective starting points. The system determines whether the sound pressure level of the presented sound is excessive to the user, based on the amount of change. 1. A sound pressure assessment system comprising:a biological signal measurement section configured to measure an electroencephalogram signal of a user;a sound database retaining data of a plurality of sounds, each sound being a pure tone;a presentation sound determination section configured to determine a sound to be presented by referring to the sound database;an output section configured to present the sound determined by the presentation sound determination section as a first sound to the user, and in a predetermined time after presenting the first sound, configured to present as a second sound a sound at least having a same frequency and a same sound pressure level as those of the first sound to the user;an amount-of-change extraction section configured to extract an amount of change from an N1 component in response to the first sound to an N1 component in response to the second sound, the N1 component in response to the first sound being a negative component of an event-related potential of the electroencephalogram signal based on a point of presenting the first sound as a starting point, the N1 component in response to ...

Apparatus and method for confirming control information of earphone in portable terminal

An apparatus and method for confirming control information that is input through an earphone in a portable terminal are provided. The apparatus includes an interface to which an earphone connects, and a controller for creating bit information dependent on a key input provided from the earphone connected through the interface, and confirming control information corresponding to the bit information.

Electrostatic parallel plate actuators whose moving elements are driven only by electrostatic force and methods useful in conjunction therewith

An actuator apparatus is provided that includes at least one actuator device, each actuator device including an array of moving elements, each individual moving element is operative to be constrained to travel alternately back and forth along a respective axis responsive to an individual first electrostatic force operative thereupon, wherein each moving element has an at-rest position and is driven away from its at rest position solely by the first electrostatic force; and at least one electrode operative to apply a controlled temporal sequence of potential differences with at least one individual moving element from among the array of moving elements thereby to selectably generate the first electrostatic force; and a controller operative to receive the digital input signal and to control at least one of the at least one electrode and the individual moving element to apply the sequence of potential differences.

SIGNAL PROCESSING SYSTEM AND METHODS FOR RELIABLY DETECTING AUDIBLE ALARMS

A signal processing system and associated methods are disclosed for reliably detecting audible alert signals, such as Temporal-3 and Temporal-4 alert signals generated by commercially available smoke, fire, and carbon monoxide detectors. The system and methods are capable of detecting audible alert signals of different intensities in the presence of dynamic background environment (e.g., television programming, music, noise, and the like). The signal processing system and methods may, in some embodiments, be incorporated into a bedside or other unit that generates a supplemental alert signal capable of alerting individuals who might not otherwise respond to the audible alert signal, such as individuals who are asleep, hearing impaired, or intoxicated. 1. A method for detecting audible alert signals , the method comprising:generating a digital representation of a received audio signal;accessing a plurality of templates stored in a memory, each template corresponding to a respective audible alert signal timing pattern, said plurality of templates encompassing an acceptable range of timing pattern variations;generating, by signal processing circuitry, a plurality of matching scores, each of which represents a degree of match between the digital representation of the received audio signal and a respective template; anddetermining, based at least partly on the matching scores, whether an audible alert signal is present.2. The method of claim 1 , wherein the plurality of templates comprises templates that correspond to variations in a Temporal-3 timing pattern.3. The method of claim 1 , wherein the plurality of templates comprises templates that correspond to variations in a Temporal-4 timing pattern4. The method of claim 1 , wherein the plurality of templates includes at least ten templates.5. The method of claim 1 , wherein generating the digital representation comprises applying band pass filtering such that the digital representation corresponds to a frequency band ...

WIRELESS DEVICE

A wireless device capable of reliably detecting a tone squelch signal in a short time is provided. The wireless device which detects a tone squelch signal contained in received baseband signals is provided with: a frequency shifting unit (quadrature demodulation unit ()) which shifts the received baseband signal in the negative frequency direction by a preset frequency of a tone squelch signal; LPF (--) for eliminating unnecessary frequency components from the output of the frequency shifting unit; and an assessment unit () which assesses presence/non-presence of the tone squelch signal on the basis of the output from the LPF (--). 1. A wireless device for detecting a tone squelch signal contained in received baseband signals , comprising:a frequency shifting unit which shifts a frequency of the received baseband signal in a negative frequency direction by a preset frequency of the tone squelch signal;a LPF which removes unnecessary frequency components from an output of the frequency shifting unit; andan assessment unit which assesses a presence or absence of the tone squelch signal based on an output of the LPF.2. The wireless device of claim 1 , wherein the frequency shifting unit shifts the frequency by performing quadrature demodulation on the received baseband signal.3. The wireless device of claim 1 , further comprising:a frequency detection unit which detects a frequency based on the output of the LPF; anda power detection unit which detects a power of the output of the LPF,wherein the assessment unit assesses that the tone squelch signal is present if the detected frequency is within a predetermined frequency range, and the detected power is equal to or greater than a predetermined power.4. A wireless device for detecting a tone squelch signal contained in received baseband signals claim 1 , comprising:a frequency shifting unit which shifts a frequency of the received baseband signal in a negative frequency direction by a preset frequency of the tone ...

METHOD AND DEVICE FOR DETECTING ACOUSTIC SHOCKS

A method and device are provided for detecting acoustic shocks in an audio stream. The method includes: breaking down the audio stream into audio frames; analyzing the audio frames in order to assign each audio frame a category value from among a plurality of predefined values; and determining the probability of an acoustic shock occurring in a current frame, based on a sequence of a given length of category values assigned to a set of frames, using a two-state Markov model, defined by a predetermined transition matrix and transmission matrix. 1. A method of detecting acoustic shocks in an audio stream , wherein the method comprises the following steps:breaking down said audio stream into audio frames;analysing said audio frames by a detection device in order to assign each audio frame a category value from a plurality of predefined values; anddetermining a probability of having an acoustic shock in a current frame by the detection device, based on a sequence of given length of category values assigned to a set of frames, using a two-state Markov model, defined by a predetermined transition matrix and transmission matrix.2. The method according to claim 1 , wherein analysing an audio frame comprises determining at least one physical indicator of the frame and discretising the at least one indicator into category values by comparison with at least one predetermined set of thresholds.3. The method according to claim 2 , wherein the physical indicator is a frequency parameter defining an energy criterion by frequency band.4. The method according to claim 3 , wherein the method also comprises a step of determining the acoustic shock frequencies as a function of the energy criterion by frequency band.5. The method according to claim 3 , wherein the energy criterion is a function of the difference between a standardised energy spectrum of the frame and a percentile function of the standardised energy spectrum.6. The method according to claim 2 , wherein the physical ...

Acoustic Shock Prevention Apparatus

An apparatus including at least one processor and at least one memory including computer program code. The at least one memory and the computer program code is configured to, with the at least one processor, cause the apparatus at least to perform: determining a change of state in an apparatus within a first time period, wherein a first state operates with a first audio output having a first average sound pressure level and a second state operates with a second audio output having a higher average sound pressure level; and processing the second audio output signal such that the second audio output sound pressure level is acoustically continuous with respect to the first average sound pressure level immediately following the change of state. 1. A method comprising:determining a change of state in an apparatus within a first time period, wherein a first state operates with a first audio output having a first average sound pressure level and a second state operates with a second audio output having a higher average sound pressure level; andprocessing the second audio output signal such that the second audio output sound pressure level is acoustically continuous with respect to the first average sound pressure level immediately following the change of state.2. The method as claimed in claim 1 , wherein processing the second audio output signal comprises at least one of:muting the second audio output for a muting time period; andramping the second audio output for an audio ramp time period.3. The method as claimed in claim 2 , wherein ramping the second audio output for an audio ramp time period comprises at least one of:linear ramping; andnon-linear ramping.4. The method as claimed in claim 1 , wherein determining a change of state in an apparatus comprises:determining an ending of the first state, wherein the first state is a member of a first group of states; anddetermining a start of the second state, wherein the second state is a member of a second group of states.5 ...

Method and Apparatus for Audio Signal Classification

An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform determining a signal identification value for an audio signal, determining at least one noise level value for the audio signal, comparing the signal identification value against a signal identification threshold and each of the at least one noise level value against an associated noise level threshold, and identifying the audio signal dependent on the comparison.

DEVICE AND METHOD FOR DIAGNOSING AUDIO CIRCUITRY

The present invention provides a device and method for diagnosing an audio circuitry, which is applied to a mobile terminal and its earphone. Both of the mobile terminal and its earphone include: a MIC circuit and a sounding circuit, wherein, this device includes: a MIC Circuit Diagnosis Module and a Sounding Circuit Diagnosis Module; A MIC Circuit Diagnosis Module is used for obtaining an amplitude and frequency of a sine wave signal sampled by the MIC circuit, determining whether the MIC circuit is qualified or not according to the amplitude and frequency and outputting the sine wave signal. A Sounding Circuit Diagnosis Module includes: a converter plate with a MIC circuit and a processor; the converter plate is used for sampling an analog signal output by the sounding circuit, converting the analog signal into a digital signal and sending it to a processor; and the processor is used for analyzing and processing the amplitude and frequency of the digital signal and determining whether the sounding circuit is qualified or not. According to the technical scheme provided by the present invention, the quality problem existing in the audio circuitry module of the mobile terminal can be solved, so as to reduce users' complaints and improve the customer satisfaction. 1. A device for diagnosing an audio circuitry , applied to a mobile terminal and its earphone , both of the mobile terminal and its earphone comprising a microphone (MIC) circuit and a sounding circuit , wherein the device for diagnosing an audio circuitry comprises:a MIC Circuit Diagnosis Module; anda Sounding Circuit Diagnosis Module;wherein the MIC Circuit Detection Module is configured for obtaining an amplitude and frequency of a sine wave signal sampled by the MIC circuit, determining whether the MIC circuit is qualified or not according to the amplitude and the frequency and outputting the sine wave signal;the Sounding Circuit Diagnosis Module comprises a converter plate with a MIC circuit and a ...

Method and apparatus for estimating interchannel delay of sound signal

A method and an apparatus for estimating an interchannel delay of a sound signal are disclosed, related to the communication field and capable of realizing a stable sound field in a crosstalk. The method includes: calculating an error between an actual interchannel phase difference and a predicted interchannel phase difference of a sound signal, where the predicted interchannel phase difference is predicted according to a predetermined interchannel delay of the sound signal; determining whether the sound signal is a sound signal in a crosstalk according to the error; and if the sound signal is a sound signal in the crosstalk, setting an interchannel delay corresponding to the sound signal to a fixed value

Method and an apparatus for a user to select one of a multiple of audio tracks

A method and apparatus for having a user select from simultaneously provided audio tracks which are provided from different directions to the user. One of the tracks is provided from in front of the user and with a higher intensity than another track which is provided from a direction more to the side of the user. When the other track is selected, it is provided from in front of the user with a higher intensity, and a third track is now provided from the side of the user and at a lower intensity.

SOUND SOURCE SEPARATION DEVICE, SOUND SOURCE SEPARATION METHOD AND PROGRAM

With conventional source separator devices, specific frequency bands are significantly reduced in environments where dispersed static is present that does not come from a particular direction, and as a result, the dispersed static may be filtered irregularly without regard to sound source separation results, giving rise to musical noise. In an embodiment of the present invention, by computing weighting coefficients which are in a complex conjugate relation, for post-spectrum analysis output signals from microphones (), a beam former unit () of a sound source separator device () thus carries out a beam former process for attenuating each sound source signal that comes from a region wherein the general direction of a target sound source is included and a region opposite to said region, in a plane that intersects a line segment that joins the two microphones (). A weighting coefficient computation unit () computes a weighting coefficient on the basis of the difference between power spectrum information calculated by power calculation units (). 1. A sound source separation device that separates , from mixed sounds containing mixed sound source signals output by a plurality of sound sources , a sound source signal from a target sound source , the sound source separation device comprising:a first beamformer processing unit that performs, in a frequency domain using respective first coefficients different from each other, a product-sum operation on respective output signals by a microphone pair comprising two microphones into which the mixed sounds are input to attenuate a sound source signal arrived from a region opposite to a region including a direction of the target sound source with a plane intersecting with a line interconnecting the two microphones being as a boundary;a second beamformer processing unit which multiplies respective output signals by the microphone pair by a second coefficient in a relationship of complex conjugate with the first coefficients ...

AUTOMATIC LOCATION-SPECIFIC CONTENT SELECTION FOR PORTABLE INFORMATION RETRIEVAL DEVICES

A method for anticipating a user's desired information using a FDA device connected to a computer network is provided. This method further comprises maintaining a database of user tendencies within the computer network, receiving sensor data from the user's physical environment via the FDA device, generating query strings using both tendency data and sensor data, retrieving data from external data sources using these generated query strings, organizing the retrieved data into electronic folders, and delivering this organized data to the user via the FDA device, In particular, a data management module anticipates the type of information a user desires by combining real time data taken from a sensor unit within a FDA and data regarding the history of that particular user's tendencies stored within the data management module. 123.-. (canceled)24. A method , comprising:a computing device detecting ambient audio information via a sensor unit;the computing device analyzing the detected audio information; andbased on the analyzing, the computing device determining information regarding an origin of the detected audio information.25. The method of claim 24 , wherein the analyzing includes performing a spectral analysis on the detected audio information.26. The method of claim 24 , wherein the detected audio information corresponds to a work of music claim 24 , and wherein the information regarding the origin of the detected audio information includes a name and artist of the work of music. This application claims priority pursuant to 35 U.S.C. §119(e) to U.S. Provisional Application No. 60/203,169, filed May 8, 2000, which application is specifically incorporated herein, in its entirety, by reference.1. Field of the InventionThe present invention relates to applications pertaining to information retrieval devices (IRD's) connected to the Internet. More specifically, this invention relates to a method and apparatus for anticipating a user's desired information and delivering ...

Method for Determining a Transmission Quality when Receiving Audio Signals and Receiving Unit

There is provided a method of determining a transmission quality when receiving audio signals which are transmitted over a frequency-modulated path in analog form. For that purpose a frequency-modulated audio signal is received and IQ demodulation of the input signal is implemented. The I- and the Q-path of the demodulated baseband signal are subjected to analog/digital conversion. Alternatively the input signal can first be digitized and then subjected to IQ demodulation. A multiplicity of N samples is detected. The amount of the N samples is formed. A reference value is determined from the amount of the N samples. Each of the N samples is standardized to the reference value. A measurement in respect of the fluctuations in the standardized amounts is determined. The measurement in respect of the fluctuations indicates the quality of the transmission path. 1. A method of determining a transmission quality when receiving audio signals which are transmitted in analog form by way of a frequency-modulated transmission path , comprising the steps:receiving a frequency-modulated audio signal; performing an IQ demodulation of the frequency-modulated audio signal and subsequent digitization of the I- and Q-signals; or', 'digitizing the received frequency-modulated audio signal and subsequent implementation of a digital IQ demodulation; and, 'a step comprising acquiring a plurality of N samples of the digitized input signals;', 'calculating the amount (B) of the N samples;', 'determining a reference value (R);', 'standardizing each of the N samples of the amount (B) to the reference value (R); and', 'determining a measurement in respect of fluctuations in the standardized amounts of the N samples;', 'wherein the measurement, in respect of the fluctuations, indicates the quality of the path., 'performing digital signal processing with the digitized signals and checking amplitude fluctuations in the received analog frequency-modulated audio signal by2. The method as set forth ...

ABNORMALITY DETECTION APPARATUS FOR PERIODIC DRIVING SYSTEM, PROCESSING APPARATUS INCLUDING PERIODIC DRIVING SYSTEM, ABNORMALITY DETECTION METHOD FOR PERIODIC DRIVING SYSTEM, AND COMPUTER PROGRAM

An abnormality detection apparatus for a periodic driving system includes a detection unit; a data obtaining unit for time series data from the detected sound; a determinism derivation unit configured to derive a plurality of values representing determinism providing an indicator of whether the time series data is deterministic or stochastic or a plurality of intermediate variations in a calculation process of the values representing determinism at a predetermined interval from the time series data; a probability distribution calculation unit. The abnormality detection apparatus further includes a determination unit configured to determine existence or non-existence of abnormality in the periodic driving system based on the probability distribution of the values representing determinism or the intermediate variations. 1. An abnormality detection apparatus for a periodic driving system which is used for an operation of a processing apparatus , comprising:a detection unit configured to detect sound from the periodic driving system;a data obtaining unit for time series data that temporally varies from the detected sound;a determinism derivation unit configured to derive a plurality of values representing determinism providing an indicator of whether the time series data is deterministic or stochastic or a plurality of intermediate variations in a calculation process of the values representing determinism at a predetermined interval from the time series data that have been obtained by the data obtaining unit;a probability distribution calculation unit configured to calculate probability distribution of the values representing determinism or the intermediate variations; anda determination unit configured to determine existence or non-existence of abnormality in the periodic driving system based on the probability distribution of the values representing determinism or the intermediate variations.2. The abnormality detection apparatus of claim 1 , further comprising: a ...

Method and apparatus for processing audio in mobile terminal

A method and an apparatus for processing audio data that controls pressure of a played sound to improve a quality of audio are provided. The method for processing audio in a mobile terminal includes: receiving the audio from a microphone; calculating by a controller a sound pressure level of the received audio; determining a sound pressure state of the received audio based on the calculated sound pressure level; and controlling an input gain of the audio based on the determination result.

ACOUSTIC WARNING LEVEL OPTIMIZATION IN AMBULATORY MEDICAL SYSTEMS

An ambulatory medical device is presented. The ambulatory medical device comprises a function module to provide the intended functionality of the device, a controller module to control the device, a sound generation module with an acoustic transducer to produce an acoustic signal and a signal generator to drive the acoustic transducer with a certain frequency. The signal generation module is arranged within a housing of the device. A tuning module varies the frequency used by the signal generator to drive the acoustic transducer and determines one or more frequencies that correspond to an optimum sound level of the acoustic signal outside of the housing of the device. The optimum sound level of the acoustic signal is a maximum sound level outside of the housing of the device, and/or a maximum perceivable sound level as determined by a user. 1. An ambulatory medical device , the ambulatory medical device comprising:a function module to provide the intended functionality of the device;a controller module to control the device;a sound generation module with an acoustic transducer to produce an acoustic signal;a signal generator to drive the acoustic transducer with a certain frequency, wherein the signal generation module is arranged within a housing of the device; anda tuning module that varies the frequency used by the signal generator to drive the acoustic transducer and determines one or more frequencies that correspond to an optimum sound level of the acoustic signal, wherein the optimum sound level of the acoustic signal is a maximum sound level outside of the housing of the device and/or a maximum perceivable sound level as determined by a user.2. The device according to claim 1 , wherein the tuning module comprises one or more sensors for obtaining feedback data directly and/or indirectly related to the sound level of the acoustic signal outside of the housing.3. The device according to claim 1 , wherein the tuning module measures an electrical claim 1 , or ...

Meta-Parameter Control for Digital Audio Data

This specification describes technologies relating to modifying digital audio data using a meta-parameter. A method is provided that includes receiving digital audio data; receiving an input modifying a meta-parameter to a particular meta-parameter value associated with two or more parameters according to a particular mapping, each of the two or more parameters being associated with the digital audio data; modifying at least a first parameter of the two or more parameters based on the particular meta-parameter value and the particular mapping such that a first meta-parameter value causes a modification of the first parameter associated with a volume control of the digital audio data and a second meta-parameter value causes a modification of a second parameter of the two or more parameters associated with dynamic range control of the digital audio data; and generating modified digital audio data from the two or more modified parameters of the digital audio data.

INFORMATION PROCESSOR, AUDIO PROCESSOR, AUDIO PROCESSING SYSTEM AND PROGRAM

An audio processor acquires processed sound data, which is obtained by processing original sound data according to a content of processing defined such that first sound is emitted under the standard sound emission condition, in association with a sound emission device, acquires a control signal, and acquires a device sound emission condition. In order to make second sound emitted by the sound emission device under the acquired device sound emission condition close to the first sound, the audio processor identifies the correction content of the processed sound data based on a difference between the device sound emission condition and the standard sound emission condition, the control signal, and the processed sound data acquired in association with the sound emission device. The audio processor corrects the processed sound data according to the correction content, and supplies the corrected processed sound data to the sound emission device. 1. An audio processor for being connected to a sound emission device that emits a sound according to supplied data , the audio processor comprising:a processed sound data acquirer that is adapted to acquire processed sound data, which is obtained by processing original sound data according to a content of processing defined such that determined first sound is emitted under a standard sound emission condition as a predetermined sound emission condition, wherein the processed sound data is acquired in association with the sound emission device;a control signal acquirer that is adapted to acquire a control signal including at least one of a parameter indicating the content of processing on the original sound data and information indicating a configuration of processed sound data obtained by processing the original sound data by means of the parameter;a sound emission condition acquirer that is adapted to acquire a device sound emission condition as a sound emission condition for the sound emission device;an identifier that is adapted ...

Speaker Impedance Measurement

An integrated circuit for processing audio and voice signals in a telephone is disclosed. The integrated circuit comprises an audio output port having a first and a second output terminal for connection to a first and a second input terminal, respectively, of a speaker. The integrated circuit further comprises an audio input port having a first and a second input terminal for connection to a first and a second output terminal of an audio input source. The integrated circuit also comprises at least one audio output path and one audio input path. Moreover, the integrated circuit comprises a test impedance, such as a resistor, (RT) for measuring an electrical impedance of the speaker. 115-. (canceled)16. An integrated circuit for processing audio and voice signals in a telephone , comprising:an audio output port having a first and a second output terminal for connection to a first and a second input terminal, respectively, of a speaker;an audio input port having a first and a second input terminal for connection to a first and a second output terminal of an audio input source;at least one audio output path having a digital input port and an analog output port with at least one analog output terminal;an audio input path having an analog input port, comprising a first and a second input terminal, and a digital output port; and{'sub': 'T', 'a test impedance, such as a resistor, (R) having a first and a second terminal, wherein the first terminal is connected to a reference-voltage node of the integrated circuit;'}wherein the integrated circuit is configured such that: one of the at least one audio output paths is arranged to supply an audio output signal to the audio output port via the analog output port of that one of the at least one audio output path; and', 'the audio input path is arranged to receive an audio input signal from the audio input port via the analog input port of the audio input path; and', 'in a measurement mode of the integrated circuit', 'the at least ...

Recording and playback device capable of repeated playback, computer-readable storage medium, and recording and playback method

In a recording and playback device of the present invention, when input data exceeding a threshold value is supplied, the CPU records input data for an amount of time corresponding to a single beat in an area specified by syllable number SPLIT in an input buffer IB of the RAM, and after incrementing the syllable number SPLIT, waits until the recorded data becomes silent. The CPU repeats this series of processing until the value of the incremented syllable number SPLIT reaches “4”, and thereby stores input data recorded for an amount of time corresponding to a single beat in each input buffer IB(1) to IB(4) corresponding to syllable numbers SPLIT1 to SPLIT4. Then, the CPU copies the input data to the recording area of the RAM such that these input data are sequentially connected and formed into loop data for an amount of time corresponding to a single bar.

DEVICE FOR DETECTING SOUNDS OUTSIDE VEHICLE

Provided is a device for detecting sound outside vehicle capable of detecting sounds outside the vehicle with high accuracy in various surrounding environments. 115.-. (canceled)16. A device for detecting sound outside vehicle comprising:a sound collecting unit that is provided in a vehicle and collects sounds outside the vehicle;a target sound detecting unit that detects a detection target sound from the sounds outside the vehicle which are detected by the sound collecting unit;a traveling road information detecting unit that detects traveling road information about a road on which the vehicle travels; anda sound collection characteristic adjusting unit that adjusts sound collection characteristics of the sound collecting unit on the basis of the detection result of the traveling road information detecting unit.17. The device for detecting sound outside vehicle according to claim 16 , further comprising:a direction detecting unit that detects the direction of the detection target sound on the basis of the detection result of the target sound detecting unit.18. The device for detecting sound outside vehicle according to claim 16 ,wherein the detection target sound is a traveling sound of an approaching vehicle which approaches the vehicle.19. The device for detecting sound outside vehicle according to claim 18 ,wherein the traveling road information detecting unit detects the shape of the road on which the vehicle travels as the traveling road information.20. The device for detecting sound outside vehicle according to claim 19 ,wherein the sound collection characteristic adjusting unit adjusts a directivity of the sound collecting unit as characteristics of the sound collecting unit on the basis of the shape of the traveling road.21. The device for detecting sound outside vehicle according to claim 20 ,wherein, when it is determined that the direction of the road on which the vehicle travels is decided on the basis of the shape of the traveling road, the sound ...

METHOD AND APPARATUS TO MEASURE DIFFERNTIAL PHASE AND FREQUENCY MODULATION DISTORTIONS FOR AUDIO EQUIPMENT

A testing method or apparatus utilizes multiple frequencies applied to a device under test for measuring newly discovered frequency modulation effects. An embodiment may include a lower frequency signal combined with two (or more) higher frequency signals to test a dynamic change in frequency response, gain, and or phase of the lower frequency signal from an audio device. This dynamic test can reveal frequency modulation effects via the two higher frequency signals emulate a modulated signal that provides a phase or frequency modulation frequency related to the difference of the two higher frequency signals. Another embodiment may include the use of a higher frequency signal and pulsed waveform to dynamically induce a time varying phase or frequency distortion of the pulsed waveform or components of the pulsed waveform from the device that has differential phase distortion. In another embodiment, the device's output is then measured with an FM detector and or phase detector to measure a phase and or frequency shift in one or more signals of the frequencies related to the pulsed waveform, such as a signal of the fundamental frequency and or one or more harmonic related to the pulsed waveform the device. 1. Method of measuring an induced phase or frequency modulation effect of an audio device , wherein the audio device has an input and an output , comprising:providing a first signal comprising a pulsed waveform and a second signal whose frequency is higher than the frequency of the first signal to an input of the audio device;wherein the output of the audio device is coupled to a Wow and Flutter meter, a phase detector, and or a frequency modulation detector;measuring the induced phase or frequency modulation effect of one or more signals related to the pulsed waveform at the output of the audio device; andwherein the phase or frequency modulation effect on one of the input signals is caused or induced by the second signal coupled to the input of the audio device.2. A ...

Hearing testing probe with integrated temperature and humidity sensors and active temperature control

Certain embodiments provide a hearing testing system. The hearing testing system includes a transducer and an environmental sensor coupled with an acoustic channel. The environmental sensor is configured to measure environmental conditions of the acoustic channel. The hearing testing system includes a processor. The processor is configured to receive the environmental conditions from the environmental sensor. The processor is configured to apply, based on the measure environmental conditions, correction data to a transducer response to generate a corrected transducer response. In certain embodiments, the processor is configured to control a heating element based on a measured temperature to maintain a pre-defined temperature, or range of temperatures, at a testing probe.

Method and System

A method of warning and/or reduction of risk of ear damage of a hearer by excessive exposure to relatively high volume and/or frequency of sound, wherein a non-audio input representative of desired sound and/or ambient sound is employed to warn of said risk and/or said risk is automatically reduced.

Loudspeaker Driver With Sensing Coils For Sensing The Position And Velocity Of a Voice-Coil

This invention concerns a loudspeaker driver includes at least one actuator connected to a vibrating support to impart excitation to the latter when caused to move, wherein the loudspeaker driver further includes a plurality of sensing members arranged to move with the at least one actuator, each sensing member providing output sensing data dependent on the velocity of said at least one actuator, and means for determining the position of the at least one actuator based on at least one ratio (X/Y) of output sensing data or of linear combinations of output sensing data provided from the plurality of sensing members, said at least one ratio being independent of the velocity of the at least one actuator. 2. The loudspeaker driver of claim 1 , wherein the position of the at least one actuator within the whole range of actuator positions is based on at least two ratios of sensing members output sensing data claim 1 , each ratio covering a portion of the whole range of actuator positions.3. The loudspeaker driver of claim 2 , wherein the given sensing member is selected according to a predetermined criterion.4. The loudspeaker driver of claim 3 , wherein the selected sensing member is the sensing member for which the ratio X/Y is substantially linear as a function of the at least one actuator position over a portion of the whole range of actuator positions.5. The loudspeaker driver of claim 1 , wherein it comprises means for determining the velocity of the at least one actuator that is axially moving within a magnetic gap of the loudspeaker driver based on the determined position of said at least one actuator and at least some of the sensing members output sensing data.6. The loudspeaker driver of claim 1 , wherein the position of the at least one actuator is determined based on at least one ratio X/Y claim 1 , where X stands for output sensing data provided by a given sensing member or by a linear combination of sensing members output sensing data and Y stands for output ...

System and Method for Determining Meter Presence Utilizing Ambient Fingerprints

Systems and methods are disclosed for providing portable device presence utilizing environmental ambient audio fingerprints. Portable devices provide media exposure data and environmental ambient fingerprints to a processing device, where the environmental ambient signatures provide at least one characteristic of the ambient audio surrounding each portable device. The environmental ambient signatures are then processed to determine if they match. Portable devices associated with matching signatures are identified. Ambient fingerprints may also be used to establish a logical location where media exposure took place. The ambient signatures are alternately combined with monitored data to provide more robust data sets for contextual processing. 1. A computer-implemented method for determining portable device presence , comprising the steps of:receiving media exposure data from a plurality of portable devices at a first input;receiving environmental ambient fingerprints from the portable devices at a second input, wherein the environmental ambient fingerprints represent at least one characteristic of an environment around each of the plurality of portable devices;processing the environmental ambient fingerprints in a processor to determine if at least two of the environmental ambient fingerprints match; andidentifying devices, via said processor, having matched environmental ambient fingerprints.2. The computer-implemented method of claim 1 , wherein the characteristic comprises a level of ambient noise over each of a series of time periods.3. The computer-implemented method of claim 2 , wherein the environmental ambient fingerprint comprises a value for each of the levels of noise claim 2 , and where in the environmental ambient fingerprint is formed using the values of the series of time periods.4. The computer-implemented method of claim 3 , wherein the value is dependent upon a determination if a level of noise in each of the series of time periods exceeds at least ...

Sensor fault detection system and method

Fault detection techniques for control of sensor systems. A sensor control integrated circuit (“IC”) may include a fault detection system for coupling to the sensor supply lines. The system may detect faults for each of the sensor supply lines. The fault detection system may level shift sensor supply line signals from a first voltage domain to a second voltage domain appropriate for the fault detection system of the controller IC. The fault detection system may level shift source potential voltages from the first voltage domain to the second voltage domain to detect predetermined fault types. The fault detection system may compare the second domain voltages from the sensor supply lines to voltages representing predetermined fault types and may generate fault status indicators based on the comparison.

REVERBERATION ESTIMATOR

A method comprising determining a reverberation time estimate for an audio signal from a first part of an encoded audio signal representing the audio signal. 1. A method comprising:determining a reverberation time estimate for an audio signal from a first part of an encoded audio signal representing the audio signal.2. The method as claimed in claim 1 , wherein determining a reverberation time estimate from an encoded audio signal further comprises:determining at least two candidate reverberation time values;determining an expectation value dependent on each of the candidate reverberation time values and the encoded audio signal; andselecting the candidate reverberation time value with the greatest expectation value as the reverberation time estimate.3. The method as claimed in claim 2 , wherein determining the expectation value comprises:determining a decay factor from the candidate reverberation time value; anddetermining a likelihood the decay factor fits the first part encoded audio encoded audio signal.4. The method as claimed in further comprising:selecting the first part of the encoded signal dependent on an activity indicator associated with the first part of the encoded signal.5. The method as claimed in claim 1 , wherein the encoded signal further comprises a second part claim 1 , each first part having an associated second part and the method further comprising at least one of:selecting the first part of the encoded signal dependent on the associated second part having a value less than a first predetermined threshold value; andselecting the first part of the encoded signal dependent on the associated second part having an average value less than a further predetermined threshold value.6. The method as claimed in claim 5 , wherein the second part is at least one of:adaptive codebook gain; andLong Term Prediction gain.7. The method as claimed in claim 1 , further comprising:determining a decaying encoded audio signal event; andselecting the first part of ...

METHODS AND SYSTEMS FOR IMPROVED ACOUSTIC ENVIRONMENT CHARACTERIZATION

In embodiments of the present invention improved capabilities are described for determining a multi-dimensional sound signature for a location within a hypothetical space, comparing the multi-dimensional sound signature to a known multi-dimensional sound signature, and modifying the hypothetical space such that the similarity between the multi-dimensional sound signature for the location within the modified hypothetical space and the known multi-dimensional sound signature is increased. 1. A method for decreasing the acoustic differences between a first room and a second room comprising: i) initiating at least one sound at a sound initiation location in the first room, and', 'ii) measuring selected characteristics of the least one sound at the sound measurement location to determine a plurality of sound reflection path characteristics of the first room between the sound initiation location and the sound measurement location;, '(a) generating a first sound signature for a sound measurement location within the first room, wherein generating the first sound signature comprises'}(b) providing a description of the second room that includes a plurality of acoustic characteristics of the second room; i) selecting a sound initiation location and the sound reception location in the second room,', 'ii) simulating a sound emission from the sound initiation location in the second room, and', 'iii) determining a plurality of sound reflection path characteristics of the second room between the sound initiation location and the sound reception location for the simulated sound emission;, '(c) calculating a second sound signature for a sound reception location within the second room, wherein calculating the second sound signature comprises'}(d) determining the differences between the first sound signature and the second sound signature;(e) adjusting the description of the second room to change at least one of the plurality of acoustic characteristics of the second room;(f) repeating ...

Wind suppression/replacement component for use with electronic systems

Techniques associated with an acoustic vibration sensor are described, including a first detector that receives a first signal and a second detector that receives a second signal and a third signal, wherein the first signal comprises a skin surface microphone signal, a static equalization filter coupled to the first detector and configured to generate an equalized first signal, a voice activity detector coupled to the first detector, and a wind detector coupled to the second detector, the wind detector configured to correlate the second signal and the third signal and to derive from the correlation a plurality of wind metrics associated with a wind noise, the wind detector is further configured to determine a magnitude associated with the wind noise, to determine whether to suspend an activity of the system, and to determine a duration of time that the magnitude associated with the wind noise exceeds a threshold.

ACOUSTIC APPARATUS

An acoustic apparatus includes: a plurality of speakers, which are connected in parallel to each other; and a line break detector for detecting the number of line break in the speakers. The line break detector includes: a detection voltage generator for applying a reference voltage to the speakers; and a line break determination device for determining the number of line break according to an actual voltage of the speakers when the reference voltage is applied to the speakers or when application of the reference voltage to the speakers is stopped. Thus, without using an impedance measurement device, the number of line break is calculated based on the actual voltage of the speakers. The number of line break in the speakers is easily detected with low manufacturing cost. 1. An acoustic apparatus comprising:a plurality of speakers, which are connected in parallel to each other; anda line break detector for detecting the number of line break in the speakers, a detection voltage generator for applying a reference voltage to the speakers; and', 'a line break determination device for determining the number of line break according to an actual voltage of the speakers when the reference voltage is applied to the speakers or when application of the reference voltage to the speakers is stopped., 'wherein the line break detector includes2. The acoustic apparatus according to claim 1 ,wherein each speaker is a capacitive speaker for generating a sound wave according to accumulation voltage change,wherein the reference voltage is a direct current voltage for saturating a total capacitance of all speakers.3. The acoustic apparatus according to claim 1 , further comprising:a digital amplifier for driving the speakers based on an input signal having a PWM waveform, which is input into the digital amplifier,wherein the detection voltage generator applies a wave-height voltage having the PWM waveform to the digital amplifier so that the digital amplifier generates the reference voltage ...

Headphone Response Optimization

Optimized sound waves presented to the listener by headphones, notwithstanding differences in ear geometry and headphone positioning. A test signal causes an acoustic sensor to receive sound waves actually formed in the listener's ear cavity. A response from the sensor is compared with an expected ear cavity transfer function, from which desired adjustments to the audio signal are determined. The audio signal might be received from an application program, calibrated by an interface software element, and adjusted thereby, before forwarding to the headphones. Calibration might be performed from when the headphones are positioned, or dynamically in response to changes in the transfer function. 1. A method , including the steps of:emitting a test sound wave from a headphone into an ear of a listener;receiving, by a sensor, a response to said test sound wave;comparing said response to an expected response to said test sound wave;determining any differences between said response and said expected response; andadjusting an input audio signal to a headphone in response to said differences.2. The method of claim 1 , wherein said expected response is responsive to one or more of a predetermined ear geometry and a predetermined headphone position.3. The method of claim 1 , including steps of:emitting a second test sound wave into a second ear of said listener;comparing a response to said second test sound wave to an expected response to said second test sound wave;determining any differences between said second response and said second expected response; andadjusting an input audio signal to said second ear in response to said differences.4. The method of claim 3 , wherein said steps of emitting a test sound wave from a headphone into an ear of a listener claim 3 , comparing said response to an expected response to said test sound wave claim 3 , and determining any differences between said response and said expected response are performed at distinct times from said steps of ...

Ultrasonic transducer device, probe, electronic instrument, and ultrasonic diagnostic device

An ultrasonic transducer device includes a substrate, a plurality of ultrasonic transducer elements, a wiring substrate and a wiring member. The substrate defines a plurality of openings arranged in an array pattern. Each of the ultrasonic transducer elements is provided in each of the openings on a first surface of the substrate. The wiring substrate is arranged to face a second surface of the substrate that is opposite from the first surface. The wiring substrate includes a first wiring part. The wiring member is connected to the substrate and the wiring substrate. The wiring member includes a second wiring part electrically connecting the ultrasonic transducer elements to the first wiring part.

LOW POWER AUDIO DETECTION

Devices and methods of detecting a predetermined audio signal in audio signals are provided. A device includes a processor coupled to a clock signal generator, a power controller and an audio detector. The power controller controls a clock rate provided to the processor by the clock signal generator, to control the device to operate in a low power mode having a relatively low power consumption or in a normal power mode having a relatively high power consumption. The audio detector receives audio signals and detects, in the low power mode, probable presence of a predetermined audio signal in the audio signals. The power controller controls the device to switch from the low power mode to the normal power mode responsive to the detected presence of the predetermined audio signal by the audio detector. 1. A method of detecting a predetermined audio signal in audio signals of a device , the device including a processor , the method comprising the steps of:controlling a clock rate provided to the processor, to operate the device in a low power mode having a relatively low power consumption;detecting, in the low power mode, probable presence of the predetermined audio signal in the audio signals; andswitching the device from the low power mode to a normal power mode responsive to the detected presence of the predetermined audio signal, by adjusting the clock rate provided to the processor, the normal power mode having a relatively high power consumption.2. The method of claim 1 , wherein the predetermined audio signal includes a voice signal or a predetermined non-voice audio signal.3. The method of claim 1 , wherein the clock rate includes a first clock rate corresponding to the low power mode and a second clock rate corresponding to the normal power mode claim 1 , the second clock rate having a higher accuracy than the first clock rate.4. The method of claim 1 , the method further including storing at least a portion of the audio signals responsive to the detecting of ...

MEASUREMENT APPARATUS AND MEASUREMENT METHOD

According to one embodiment, a measurement apparatus is configured to supply, to a output device, a second measurement signal to sweep the frequency, to receive a second received signal output from the output device when the second measurement signal is supplied to the output device, and to calculate an impulse response by convolving the second received signal and an inverted measurement signal having an inverted characteristic of the second measurement signal. The second measurement signal has a signal characteristic obtained by multiplying the first measurement signal by a characteristic of a characteristic of the amplitude spectrum. 1. A measurement apparatus configured to measure an impulse response by using an object system comprising an output device configured to output a first output signal corresponding to a supplied first measurement signal for sweeping a frequency , and a receiving device configured to output a first received signal by receiving the first output signal , the first received signal having a known characteristic of an amplitude spectrum , the apparatus comprising:an output module configured to output, to the output device, a second measurement signal obtained by multiplying the first measurement signal by a weight comprising an inverted characteristic of the amplitude spectrum;a receiving module configured to receive a second received signal output from the receiving device when the second measurement signal is supplied from the output module to the output device; andan impulse response calculator configured to calculate the impulse response by convolving the second received signal and an inverted measurement signal having an inverted characteristic of the second measurement signal.2. The apparatus of claim 1 , wherein the first measurement signal comprises a sine sweep signal whose frequency sequentially varies with time.3. The apparatus of claim 2 , wherein the sine sweep signal comprises a time-stretched pulse (TSP) signal.4. The ...

Piezoelectric Speaker Device

A piezoelectric speaker device that includes an insulation layer having an electric insulation property formed on a user-side driving electrode, and flaw detection electrode lines formed on the insulation layer. If the flaw detection electrode lines are determined to be damaged due to the occurrence of a flaw, the driving voltage applied to the driving electrodes is lowered. In order to prevent electric-shock accidents, the driving voltage is lowered to below 42.4 V, preferably, and to 0 V, more preferably. 1. A piezoelectric speaker device comprising:an organic polymer piezoelectric film;at least one pair of driving electrodes provided in contact with the piezoelectric film and configured to apply a driving voltage to drive the piezoelectric film, the at least one pair of driving electrodes including a user-side driving electrode on a first side of the piezoelectric film;an insulation layer on the user-side driving electrode;a flaw detection electrode line on the insulation layer;a detection circuit configured to detect whether the flaw detection electrode line is in a normal electric conduction state; anda driving-voltage control circuit configured to control the driving voltage applied to the at least one pair of driving electrodes based on a result of detection by the detection circuit.2. The piezoelectric speaker device according to claim 1 , further comprising a protective layer on the insulation layer and overlaying the flaw detection electrode line.3. The piezoelectric speaker device according to claim 2 , wherein the protective layer is made of a resin selected from the group consisting of polyethylene terephthalate claim 2 , polyethylene naphthalate claim 2 , polymethylmethacrylate claim 2 , polycarbonate and polypropylene.4. The piezoelectric speaker device according to claim 1 , further comprising a notification circuit configured to notify a user when the detection circuit has recognized damage in the flaw detection electrode line.5. The piezoelectric ...

System and method for mapping and displaying audio source locations

The present invention relates generally to a method and system for defining a reference sound position and producing an indicia proximate thereto in relation to one or more sound characteristics. The present invention, in one or more embodiments, provides for displaying a holographic image at a reference sound location which is determined in relation to the identification of one or more target sounds being associated with one or more identified sound characteristics. In other embodiments, the present invention provides for an indicia to appear to be interactive with a reference sound location and may be used in a variety of environments including but not limited to rock theatrics, homeland security and residential security. 1. A method for defining a reference sound position and producing an indicia proximate thereto in relation to one or more sound characteristics at a predetermined location , comprising:defining at least one sound characteristic to be detected;detecting at least one target sound in relation to the at least one sound characteristic;determining the referenced sound position in relation to the detected target sound; and,producing the indicia proximate to the determined referenced sound position.2. The method of claim 1 , wherein the sound characteristics are one or more of:frequency range, decibel level, pitch range, loudness range, directional location, and period of time.3. The method of claim 2 , wherein the referenced sound position includes a two-dimensional position in relation to the predetermined location frame.4. The method of claim 3 , wherein the referenced sound position includes a three-dimensional position in relation to the predetermined location frame.5. The method of claim 2 , wherein the at least one target sound is detected using one or more of a soundwave detector claim 2 , acoustic camera claim 2 , or electronic control logic.6. The method of claim 5 , wherein the soundwave detector is one of a(n): microphone claim 5 , electronic ...

INTEGRATED MOTION DETECTION USING CHANGES IN ACOUSTIC ECHO PATH

A system for detecting motion comprising a first speaker, a first microphone separated from the first speaker by a distance D, a sound generator, an echo parameter measurement device and an echo parameter monitor, wherein the echo parameter monitor stores two or more sequential echo parameters and generates a motion indicator if the two or more sequential echo parameters indicate a change in an acoustic echo path that exceeds a predetermined threshold. 1. A system for detecting motion comprising:a first speaker;{'b': '1', 'a first microphone separated from the first speaker by a distance D;'}a sound generator;an echo parameter measurement device; andan echo parameter monitor, wherein the echo parameter monitor stores two or more sequential echo parameters and generates a motion indicator if the two or more sequential echo parameters indicate a change in an acoustic echo path that exceeds a predetermined threshold.2212. The system of further comprising a second speaker separated from the first microphone by a distance D claim 1 , wherein D is different from D.3212. The system of further comprising a second speaker separated from the first microphone by a distance D claim 1 , wherein D is substantially equal to D.4212. The system of further comprising a second microphone separated from the first speaker by a distance D claim 1 , wherein D is different from D.5212. The system of further comprising a second microphone separated from the first speaker by a distance D claim 1 , wherein D is substantially equal to D.6. The system of further comprising:{'b': 2', '1', '2, 'a second speaker separated from the first microphone by a distance D, wherein D is different from D; and'}{'b': 3', '3', '1', '2, 'a second microphone separated from the first speaker by a distance D, where D is different from D and D.'}7. The system of further comprising:{'b': 2', '1', '2, 'a second speaker separated from the first microphone by a distance D, wherein D is substantially equal to D; and'}{' ...

EARHEALTH MONITORING SYSTEM AND METHOD IV

A system for monitoring sound pressure levels at the ear includes an ambient sound microphone (ASM) for receiving ambient sounds and an ear canal microphone (ECM) for producing audio signals as a function of ambient sound received at the ambient sound microphone and a sound signal received from an associated personal audio device. A logic circuit is operatively associated with the ASM and ECM calculates a total SPL_Dose experienced by the ear at a time t. The total SPL_Dose is calculated by determining SPL_Dose for periods Δt as measured at the ECM. The logic circuit may select an action parameter in response to the Total SPL_Dose. 1. A system configured to monitor a sound pressure level dose at an ear comprising:an ear canal microphone configured to measure an acoustic sound pressure level (SPL) within an ear canal of the ear; anda logic circuit operatively coupled to the ear canal microphone, where the logic circuit calculates a Total SPL_Dose at a time t as a function of the SPL.2. The system of claim 1 , wherein the logic circuit calculates the Total SPL_Dose as a function of an estimated SPL_Dose during a time period Δt.3. The system of claim 2 , wherein the estimated SPL_Dose is calculated as: SPL=SPL_Dose+Time of Sound Exposure/Time 100% claim 2 , where Time 100%=24 hrs/2̂((L−T)/3) claim 2 , where Lis a measured Ear Canal d/BA SPL by the ear canal microphone claim 2 , and where T is a threshold value related to noise level.4. The system of claim 2 , further comprising a readable memory operatively connected to the logic circuit claim 2 , at least one action parameter stored in the readable memory claim 2 , the logic circuit comparing the Total SPL_Dose to a threshold value claim 2 , and if the Total SPL_Dose is greater than the threshold value claim 2 , then the logic circuit reads the action parameter from the readable memory.5. The system of claim 4 , wherein the logic circuit performs an action associated with the action parameter claim 4 , where the ...

ANALYSIS SYSTEM AND METHOD FOR AUDIO DATA

An analysis system and method for audio data related to a user is provided, so that the user can be classified as one of multiple classes with an assumed probability based on the analysis result. The analysis system comprises an audio transformer () adapted to transform the audio data related to the user into spectra data; a pattern recognizer () adapted to decompose the spectra data to predetermined eigenvectors to get the decomposition pattern of the spectra data; a scorer () adapted to calculate the assumed scores of the multiple classes related to the user based on the decomposition pattern of the spectra data and the attributes of the user using a trained model. 1. An analysis system for analysis of audio data related to a user , comprising:an audio transformer adapted to transform the audio data into a spectra data;a pattern recognizer adapted to decompose the spectra data to predetermined eigenvectors to get a decomposition pattern of the spectra data; anda scorer adapted to calculate assumed scores of multiple classes related to the user based on the decomposition pattern of the spectra data and attribute of the user using a trained model.2. The audio analysis system according to claim 1 , wherein the scorer is adapted to attribute the user to a class with highest assumed score among all of the multiple classes.3. The audio analysis system according to claim 1 , further comprising:a trainer adapted to train the trained model based on at least one history item each comprising a decomposition pattern of a spectra data corresponding to a history audio data of a history user, attributes of the history user, and an actual score of one of the multiple classes for the history user.4. The audio analysis system according to claim 3 , wherein the trainer is adapted to retrain the trained model based on the history items and a new item comprising the decomposition pattern of the spectra data claim 3 , the attributes of the user claim 3 , and an actual score of an ...

Method and system of equalization pre-preocessing for sound receivng system

An exemplary embodiment illustrates an equalization pre-processing method, adapted for characterizing a second sound receptor unit in a sound receiving system based on knowing the internal structure parameters of a first sound receptor unit. The method includes: measuring a first sensitivity response of the first sound receptor unit and a second sensitivity response of the second sound receptor unit; equalizing the second sensitivity response according to the first sensitivity response and obtaining the differences in the sensitivity response; conducting simulation to obtain the third sensitivity response associated with the first sound receptor unit in the given sound receiving system; compensating the third sensitivity response to generate the fourth sensitivity response associated with the second sound receptor unit in the sound receiving system according to the differences in the sensitivity response; analyzing the fourth sensitivity response to characterize the second sound receptor unit in the given sound receiving system.

Audio signal measurement method for speaker and electronic apparatus having the speaker

An audio signal measurement method for a speaker and an electronic apparatus having the speaker are provided. The electronic apparatus further has a processing circuit and a power amplifier. The processing circuit is coupled to the speaker and configured to execute a time domain to frequency domain transform according to a voltage value of an audio signal and a current value of current feedback from the speaker so as to obtain a frequency response curve. The power amplifier is coupled to the speaker and configured to drive the speaker according the voltage value of the audio signal. The processing circuit is capable of determining whether the frequency response curve is located within a predetermined area such that the processing circuit generates a signal when the frequency response curve is located out of the predetermined area. Thereby, the electronic apparatus may measure its transducer distortion and acoustic box leakage.

System for Detecting and Reducing Noise via a Microphone Array

A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment. 1. A computer-implemented method for detecting noise , comprising the steps of:receiving a plurality of signals from a microphone array;in a first computer process, decomposing the signals into frequency sub-band signals;in a second computer process, determining time dependent measures for the signals based on the frequency sub-band signals;in a third computer process, evaluating a predetermined criterion function using the time dependent measures, producing a criterion result that determines if noise is present within the signals; andin a fourth computer process, using results of the evaluating to detect noise according to a predetermined threshold;wherein the criterion function is a variance of the time dependent measure.2. The method of where the microphone array further comprises at least two microphones.3. The method of where the predetermined criterion function is a predetermined statistical function of the time dependent measures.4. The method of where the decomposing step further comprises digitizing each signal in the plurality of signals and decomposing each digitized signal into a complex-valued frequency sub-band signal using short time discrete Fourier transform.5. The method of where the decomposing step further comprises digitizing each signal in the plurality of signals and decomposing each digitized signal into a complex-valued frequency sub-band signal using a discrete Wavelet transform.6. The method of where the decomposing step further comprises digitizing each signal in the plurality of ...

CUSTOM IN-EAR MONITOR

An in-ear monitor that can be customized for particular applications and individuals includes a housing formed from a body and a cover. A dynamic driver is mounted in a cavity in the housing on an angled mounting flange. The dynamic driver is acoustically coupled to a trumpet-shaped sound collector. The trumpet-shaped sound collector is coupled to a main sound bore that exits an opening in a nozzle portion of the body that is inserted into the ear canal of a user. An ambient sound port collects ambient sound and couples it to the sound bore. An additional bass post increases the bass response of the monitor. Ear impressions are used to customize the body of the monitor to the ear of a user and the location of the bass and ambient sound ports can be altered for different applications. 1. An in-ear monitor , said in-ear monitor comprising:a housing having a cover and a body;a trumpet-shaped sound collector;a main sound bore acoustically coupled to said trumpet-shaped sound collector;a nozzle having a nozzle opening in said body acoustically coupled to said main sound bore;a bass sound port acoustically coupled to said nozzle opening;an ambient sound port acoustically coupled to said nozzle opening; anda dynamic driver acoustically coupled to said trumpet shaped sound collector.2. The in-ear monitor of wherein said bass sound port further comprises a bass port valve that selectively restricts a sound flow through said bass sound port.3. The in-ear monitor of wherein said ambient sound port further comprises an ambient port valve that selectively restricts a sound flow through said ambient sound port.4. The in-ear monitor of wherein said nozzle includes a recessed channel and a sealing o-ring positioned on a tip of said nozzle.5. The in-ear monitor of further comprising a recessed cable connector.6. The in-ear monitor of further comprising a balanced armature receiver positioned inside said main sound bore.7. The in-ear monitor of wherein said dynamic driver further ...

AUDIO MONITORING DEVICE

An audio monitoring device for testing an audio device is provided. The audio device is placed in a shielding room which contains a signal emitter. The signal emitter emits electromagnetic waves which electronically interfere with the audio device. The audio monitoring device includes a sound collector, a sound transmitting tube, and a sound outputting element. The sound collector is positioned in the shielding room and collects sounds from the audio device produced whilst being subjected to the electromagnetic interference issued by the signal emitter. The sound transmitting tube has one end connected to the sound collector and the other end exposed from the shielding room, to connect to the sound outputting element. The sound transmitting tube transmits the sounds collected by the sound collector to the sound outputting element for analysis. 1. An audio monitoring device , the audio monitoring device being for testing an audio device which is placed in a shielding room having a signal emitter , the signal emitter emitting electromagnetic waves to interfere with the audio device , the audio monitoring device comprising:a sound collector positioned in the shielding room, the sound collector collecting sounds emitted by the audio device while the audio device being subjected to the interfering electromagnetic waves of the signal emitter;a sound outputting element; anda sound transmitting tube having one end connecting to the sound collector and the other end exposing from the shielding room to connect the sound outputting element, the sound transmitting tube transmitting the sounds collected by the sound collector to the sound outputting element for testing.2. The audio monitoring device as claimed in claim 1 , wherein the sound collector has an open claim 1 , trumpet horn-shaped end claim 1 , the open end is aimed at the audio device to collect the sounds claim 1 , the sound collector is non-metallic.3. The audio monitoring device as claimed in claim 2 , wherein the ...

LOUDSPEAKER DRIVE CIRCUIT FOR DETERMINING LOUDSPEAKER CHARACTERISTICS AND/OR DIAGNOSTICS

A loudspeaker amplifier drive circuit performs analysis of various signals relating to the input and output of the amplifier such that the characteristics and/or diagnostics of a loudspeaker driven by the amplifier can be derived. These are then presented as outputs, so that different circuitry can make use of the information for audio signal processing. 1. A loudspeaker drive circuit for deriving at least one of characteristics and diagnostics of a loudspeaker driven by the drive circuit , comprising:an amplifier;an element for determining a clipping level of the amplifier;an element for sensing the current flowing into the loudspeaker;a processing module to determine the loudspeaker characteristics and/or diagnostics, based on an input audio, the sensed current and the clipping level of the amplifier; andan element for outputting the characteristics and/or diagnostics.2. A circuit as claimed in claim 1 , wherein the processing module is adapted to derive a loudspeaker temperature.3. A circuit as claimed in claim 1 , wherein the processing module is adapted to derive the amplifier output voltage which is supplied to the loudspeaker based on the audio input provided to the circuit and the known gain and clipping level of the amplifier.4. A circuit as claimed in claim 3 , wherein the drive circuit comprises a delay element between the audio input and the amplifier claim 3 , such that the derived amplifier output voltage is a predicted amplifier output voltage to be supplied to the loudspeaker after a delay of the delay element.5. A circuit as claimed in claim 3 , wherein the processing module is adapted to derive an electrical impedance function from the amplifier output voltage and the sensed current.6. A circuit as claimed in claim 5 , wherein the processing module is adapted to derive a voltage to displacement transfer function for the loudspeaker claim 5 , and thereby derive the loudspeaker excursion based on the amplifier output voltage.7. A circuit as claimed ...

TEST DEVICE FOR A SPEAKER MODULE FOR A LISTENING DEVICE

The application relates to a test device for a speaker module of a listening device, the speaker module comprising a speaker unit for converting an electric output signal to an output sound. The application further relates to a test system and to a data update device. An object of the present application is to provide an alternative scheme for identifying a receiver (speaker) in a listening device, e.g. a hearing aid. The problem is solved in that the test device being a separate device adapted for being electrically connected to the speaker module in a test situation, but not during normal operation of the listening device. An advantage of the invention is that it provides a flexible alternative to a receiver identification solution that is integrated into the listening device. 1. A test device for a speaker module of a listening device , the speaker module comprising a speaker unit for converting an electric output signal to an output sound , ["a communication interface to a programming device for modifying parameters of said listening device according to a user's needs, and/or", 'a communication interface to a network for exchanging data with a server connected to the network., 'the test device being a separate device adapted for being electrically connected to the speaker module in a test situation, but not during normal operation of the listening device, and the test device comprises'}2. A test device according to comprising a measurement unit for performing a measurement contributing to a characterization of the speaker module.3. A test device according to configured to identify a type of speaker unit.4. A test device according to wherein the measurement unit is adapted to measure an impedance of the speaker unit.5. A test device according to wherein the measurement unit is adapted to perform a measurement involving a separate electronic identification component of the speaker module.6. A test device according to wherein the separate electronic identification ...

REAL EAR MEASUREMENT SYSTEM USING THIN TUBE

An embodiment of a hearing assistance apparatus for performing a Real Ear Measurement (REM), comprises a hearing assistance device housing, a microphone within the housing, an earhook connected to the housing, and a flexible tube. The house has a first opening for guiding sound into the housing to the microphone. The housing and the connected earhook form an interface, where the earhook has a shape to provide a slot near the interface of the housing and the earhook. The tube guides sound, and has a first end and a second end. The first end of the flexible tube and the slot of the earhook cooperate to retain the first end of the flexible tube in the slot of the earhook and flush with the housing to provide a sound-tight connection with the first opening. 1. A method for performing a Real Ear Measurement (REM) for a user's canal using a hearing assistance apparatus with a receiver , a microphone , and a sound tube , comprising:presenting a periodic signal to the receiver to provide a calibrated sound in the user's ear canal;using the microphone and the sound tube to capture a plurality of samples from the sound in the ear canal for each desired frequency;producing a real-ear coupler difference (RECD) using the plurality of samples and a coupler response; andstoring the RECD in memory of the hearing assistance device.2. The method of claim 1 , further comprising transforming the samples into a frequency domain claim 1 , and checking for bad capture for each sample.3. The method of claim 1 , further comprising transforming the samples into a frequency domain claim 1 , and checking for body movements.4. The method of claim 1 , further comprising transforming the samples into a frequency domain claim 1 , and generating an average on the samples.5. The method of claim 1 , further comprising:transforming the samples into a frequency domain;checking for temporal variations for each sample while in the frequency domain to find clean samples; andgenerating an average using ...

AMBIENT INFORMATION NOTIFICATION APPARATUS

An ambient information notification apparatus includes: an interior sound acquisition device acquiring an interior sound in a compartment of the vehicle; an ambient information presentation sound generator generating an ambient information presentation sound; and an ambient information presentation sound output device outputting the ambient information presentation sound. The ambient information presentation sound satisfies that a sound pressure level of the ambient information presentation sound is higher than the interior sound in a predetermined frequency band, and is lower than or equal to the interior sound in other frequency band, and that the ambient information presentation sound is provided by stereophony, in which a sound image localization direction approximately directs to a virtual sound source. 1. An ambient information notification apparatus for a vehicle comprising:an interior sound acquisition device that acquires a sound in a compartment of the vehicle, which is referred as an interior sound;an ambient information presentation sound generator that generates an ambient information presentation sound, which satisfies a first condition and a second condition with regard to the interior sound; andan ambient information presentation sound output device that outputs the ambient information presentation sound,wherein the first condition is that a sound pressure level of the ambient information presentation sound is higher than the interior sound in a predetermined frequency band, and is lower than or equal to the interior sound in other frequency band, andwherein the second condition is that the ambient information presentation sound is provided by stereophony, in which a sound image localization direction approximately directs to a virtual sound source.2. The ambient information notification apparatus according to claim 1 , further comprising:an exterior object detector that detects an exterior object disposed on an outside of the vehicle,wherein the ...

HEARING DEVICE WITH AN INERTIAL MEASUREMENT UNIT

A new hearing device is provided with an inertial measurement unit for determination of yaw of the orientation of a user's head, when the hearing device is worn in its intended operational position on the user's head. 1. A hearing device configured to be head worn for emission of sound towards at least one of the ears of a user and accommodating an inertial measurement unit positioned for determining head yaw , when the user wears the hearing device in its intended operational position on the user's head.2. The hearing device according to claim 1 , wherein the inertial measurement unit comprises a first gyroscope for determining the head yaw.3. The hearing device according to claim 2 , wherein the inertial measurement unit comprises a second gyroscope for determining head pitch claim 2 , when the user wears the hearing device in its intended operational position on the user's head.4. The hearing device according to claim 1 , wherein the inertial measurement unit comprises at least one accelerometer for determining displacement of the user claim 1 , when the user wears the hearing device in its intended operational position on the user's head.5. The hearing device according to claim 4 , wherein the inertial measurement unit comprises a plurality of accelerometers with mutually orthogonal measurement axes.6. The hearing device according to claim 1 , wherein the inertial measurement unit comprises an electronic compass.7. The hearing device according to claim 1 , wherein the inertial measurement unit comprises a GPS unit for determining the geographical position of the user claim 1 , when the user wears the hearing device in its intended operational position on the user's head.8. The hearing device according to claim 1 , comprising a processor that is connected with the inertial measurement unit and configured to process data from sensor(s) of the inertial measurement unit to determine head yaw value(s).9. The hearing device according to claim 8 , wherein the processor ...

APPARATUS AND METHOD FOR INTERFACING EARPHONE

An apparatus and method for interfacing with an earphone are provided. The method includes checking if the earphone is uninstalled promptly after the connection of the earphone is sensed in an external port and if it is sensed that the earphone is uninstalled, continuing to apply a voltage to a MIC_BIAS terminal. The generation of a noise of the MIC_BIAS terminal caused by applying of a periodic voltage resulted from loose insertion of the earphone is prevented. 1. An apparatus for interfacing with an earphone , the apparatus comprising:an external port configured to interface with the earphone; andat least one processor configured to control to apply a voltage to a MIC_BIAS terminal if connection of the earphone to the external port is sensed, and continue to apply the voltage to the MIC_BIAS terminal when disconnection of the earphone is sensed.2. The apparatus of claim 1 , further comprising an output means configured to notify a user of loose insertion of the earphone through the processor.3. The apparatus of claim 2 , wherein the output means is a display unit configured to display an error message indicating the loose insertion of the earphone to the user through the processor.4. The apparatus of claim 2 , wherein the output means is a sound output means configured to notify claim 2 , by means of a sound claim 2 , the user of the loose insertion of the earphone through the processor.5. The apparatus of claim 2 , wherein the processor configured to set a timer with a predetermined time when disconnection of the earphone is sensed claim 2 , and drive the output means to notify the user it when the timer has expired.6. The apparatus of claim 5 , wherein after driving the output means claim 5 , the processor is configured to release the output means upon one of a lapse of the predetermined time claim 5 , sensing user's key manipulation and sensing normal installation or uninstallation of the earphone.7. The apparatus of claim 1 , wherein the processor is ...

Earphone

The earphone of the Present Disclosure includes a driver for converting electric signals into sound waves, a cord connected to the driver, an elastic holder, a housing and an earpiece. The elastic holder integrally possesses a cylindrical driver housing portion for inserting the driver, and a cord housing portion for housing the end portion of the cord. The housing possesses a cylindrical main body for inserting at least the driver housing portion of the elastic holder, and a sound conduit provided in the end portion of the main body. The earpiece is fitted into the ear canal.

Electronic Devices for Reducing Acoustic Leakage Effects and Related Methods and Computer Program Products

An electronic device includes a housing comprising a speaker configured to be positioned adjacent an ear of a user, and a position-sensitive region on the housing that is configured to sense a position of the ear of the user relative to the housing when the ear is positioned adjacent the speaker and to generate an acoustic leakage position signal responsive to the sensed position of the ear of the user. A control unit is in communication with the speaker and the position-sensitive region and is configured to provide an electrical input signal to the speaker. The control unit is further configured to receive the acoustic leakage position signal and to adapt the electrical input signal responsive to the acoustic leakage position signal.

Sound pickup device and imaging device

A sound pickup device includes a detector configured to detect attachment of an external apparatus, a sound pickup unit configured to pick up a sound and output a sound signal, and a noise processor configured to, in response to the detection of attachment of an external appliance, add a noise signal to the sound signal output from the sound pickup unit and output the sound signal including the noise signal added thereto.

In-Ear Audio Device Customization

An in-ear audio device has a casing on which is disposed one or more bags are positioned to be filled with fillable material during a customization process in which a test sound is acoustically output into an ear canal by an acoustic driver of the in-ear audio device while the one or more bags are being filled, and a microphone acoustically coupled to the ear canal is employed to detect sounds within the ear canal that are indicative of the frequency response of the acoustic output of the acoustic driver to determine when the degree of sealing of the ear canal by the one or more bags is sufficient to achieve a desired quality of frequency response. 1. An apparatus comprising:a casing shaped to fit into a portion of an ear, the casing comprising a bag fillable with a filling material to customize the fit of the casing within the portion of the ear to create a seal between the casing and the portion of the ear that acoustically separates an ear canal of an ear from an external environment that is external to the ear and casing;an acoustic driver disposed within the casing to acoustically output a test sound into the ear canal; andat least one aperture formed in the casing to permit a portion of a microphone to be inserted therethrough from the external environment external and into the ear canal to acoustically couple the microphone to the ear canal.2. The apparatus of claim 1 , wherein:structure of the casing forms an open interior portion within the casing that is interposed between the ear canal and the external environment, and that separates the open interior portion from the ear canal and the external environment;the at least one aperture formed in the casing comprises a first aperture coupling the open interior portion to the external environment and a second aperture coupling the open interior portion to the ear canal; andthe first and second apertures are sufficiently aligned to enable the portion of the microphone to be inserted through both the first and ...

METHOD AND SYSTEM FOR SOUND MONITORING OVER A NETWORK

A mobile communication environment () can include a mobile device () to measure and send sound pressure level data. The mobile device () can initiate the collection of audio information responsive to detecting a trigger event. Mobile device () can measure or calculate the sound pressure level from the audio information. Metadata including time information and geographic location information can be captured with the collected audio information. Mobile device () can send the sound pressure level data and metadata through a wired or wireless communication path to a database (). 1. A sound pressure level measurement system comprising:a plurality of mobile devices each having a sound pressure level measurement for providing to a database; anda communication path for the plurality of mobile devices to couple to the database where a mobile device opens a communication path to the database and uploads the sound pressure level measurement to the database.2. The sound pressure level measurement system as recited in where the communication path comprises a radio frequency communication network having at least one base receiver for coupling the plurality of mobile devices to the database.3. The sound pressure level measurement system as recited in where the communication path comprises a wireless local area network coupling the mobile device having one or more sound pressure level measurements to the database for providing the sound pressure level measurements thereto.4. The sound pressure level measurement system as recited in where the communication path comprises a wired or fiber optic path coupling a mobile device having one or more sound pressure level measurements to the database for providing the sound pressure level measurements thereto.5. The sound pressure level measurement system as recited in where the plurality of mobile devices measure sound pressure levels automatically and where metadata is provided with each sound pressure level measurement including time ...

Segmenting Audio Signals into Auditory Events

In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal, calculating the difference in spectral content between successive time blocks of the audio signal, and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold. In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events. Optionally, the invention may also assign a characteristic to one or more of the auditory events. Auditory events may be determined according to the first aspect of the invention or by another method. 1. A method for processing a multi-channel audio signal , the method comprising:analyzing time concurrent samples of the multi-channel audio signal across at least two channels to determine auditory event boundaries in the at least two channels;identifying auditory event boundaries in the at least two channels, wherein an auditory event is between adjacent auditory event boundaries, and each auditory event boundary represents an end of a preceding auditory event and a beginning of a next auditory event; andidentifying a combined auditory event boundary for the multi-channel audio signal in response to the identifying auditory event boundaries in the at least two channels, wherein neither auditory event boundaries nor auditory events are known in advance of the identifying auditory event boundaries in the at least two channels of the multi-channel audio signal.2. The method of wherein at least one of the auditory event boundaries divides: (i) a portion of audio signal quiet relative to a ...

Displaying Sound Indications On A Wearable Computing System

Example methods and systems for displaying one or more indications that indicate (i) the direction of a source of sound and (ii) the intensity level of the sound are disclosed. A method may involve receiving audio data corresponding to sound detected by a wearable computing system. Further, the method may involve analyzing the audio data to determine both (i) a direction from the wearable computing system of a source of the sound and (ii) an intensity level of the sound. Still further, the method may involve causing the wearable computing system to display one or more indications that indicate (i) the direction of the source of the sound and (ii) the intensity level of the sound. 1. A computer-implemented method comprising:receiving audio data corresponding to sound detected by a wearable computing system, wherein the wearable computing system provides a view of a real-world environment of the wearable computing system;analyzing the audio data to determine (i) a direction from the wearable computing system of a source of the sound and (ii) an intensity level of the sound;determining that the source of the sound is located either outside the view or in a periphery of the view; andcausing the wearable computing system to display one or more indications that indicate (i) the direction of the source of the sound and (ii) the intensity level of the sound, wherein the one or more indications comprise a flashing light in the periphery of the view.2. The method of claim 1 , wherein the intensity level of the sound varies with time claim 1 , and wherein the one or more indications comprise an indication associated with the intensity level that varies in phase with the intensity level.3. The method of claim 1 , wherein displaying the one or more indications comprises displaying different colors for different intensity levels of the sound.4. The method of claim 1 , wherein the one or more indications comprise an intensity meter.5. A non-transitory computer readable medium ...

Scene Change Detection Around a Set of Seed Points in Media Data

Techniques for scene change detection around seed points in media data are provided. Media features of many different types may be extracted from the media data. One or more statistical patterns of media features in a plurality of time-wise intervals around a plurality of seed time points of the media data may be determined using one or more types of features extractable from the media data. At least one of the one or more types of features comprises a type of features that captures structural properties, tonality including harmony and melody, timbre, rhythm, loudness, stereo mix, or a quantity of sound sources as related to the media data. A plurality of beginning scene change points and a plurality of ending scene change points in the media data may be detected, based on the one or more statistical patterns, for the plurality of seed time points in the media data. 1. A method for scene change detection in media data , comprising:determining one or more statistical patterns of media features in a plurality of time-wise intervals around a plurality of seed time points of the media data using one or more types of features extractable from the media data, at least one of the one or more types of features comprising a type of features that captures structural properties, tonality including harmony and melody, timbre, rhythm, loudness, stereo mix, or a quantity of sound sources as related to the media data;detecting, based on the one or more statistical patterns, a plurality of beginning scene change points and a plurality of ending scene change points in the media data for the plurality of seed time points in the media data;wherein the method is performed by one or more computing devices.2. The method of claim 1 , wherein the plurality of seed time points comprises at least one seed time point located inside a repeating section of the media data.3. The method of claim 1 , wherein the plurality of seed time points comprises at least one seed time point located inside a ...

ESTIMATION AND SUPPRESSION OF HARMONIC LOUDSPEAKER NONLINEARITIES

A harmonic echo power estimator estimates power of echo generated by harmonic loudspeaker nonlinearities in a user equipment having an echo path between a loudspeaker input and a microphone output. The estimator includes a frequency band mapper that maps each frequency band in a set of loudspeaker output signal frequency bands into a corresponding array of loudspeaker input signal frequency bands, where each frequency band in the set is mapped into several frequency bands in the corresponding array. A power estimator determines a power estimate of each input signal in each array of frequency bands. A power estimate combiner combines determined power estimates in each array of frequency bands into a corresponding estimate of loudspeaker input power generating harmonic loudspeaker nonlinearities. A power estimate transformer transforms the estimates of loudspeaker input power across the echo path into power estimates of the echo generated by the harmonic loudspeaker nonlinearities. 1. A method of estimating power of echo generated by harmonic loudspeaker nonlinearities in a user equipment having an echo path between a loudspeaker input and a microphone output , said method comprising the steps of:{'sub': lsp', 'lsp, 'mapping each frequency band in a set of loudspeaker output signal frequency bands (b) into a corresponding array of loudspeaker input signal frequency bands (b(b, k)), each frequency band in the set being mapped into several frequency bands in the corresponding array;'}{'sub': x', 'lsp, 'determining a power estimate ({circumflex over (P)}(b(b, k))) of each input signal in each array of frequency bands;'}{'sub': x', 'lsp', 'x,nl', 'lsp, 'combining determined power estimates ({circumflex over (P)}(b(b, k))) in each array of frequency bands into a corresponding estimate ({circumflex over (P)}(b)) of loudspeaker input power generating harmonic loudspeaker nonlinearities; and'}{'sub': x,nl', 'lsp', '{tilde over (s)},nl, 'transforming the estimates ({circumflex ...

User Interfaces for Controlling and Manipulating Groupings in a Multi-Zone Media System

In general, user interfaces for controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a user interface is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network. The user interface is further configured to illustrate graphically a size of a group, the larger the group appears relatively, the more plays there are in the group.

Audio User Interaction Recognition and Context Refinement

A system which tracks a social interaction between a plurality of participants, includes a fixed beamformer that is adapted to output a first spatially filtered output and configured to receive a plurality of second spatially filtered outputs from a plurality of steerable beamformers. Each steerable beamformer outputs a respective one of the second spatially filtered outputs associated with a different one of the participants. The system also includes a processor capable of determining a similarity between the first spatially filtered output and each of the second spatially filtered outputs. The processor determines the social interaction between the participants based on the similarity between the first spatially filtered output and each of the second spatially filtered outputs. 1. A system which tracks a social interaction between a plurality of participants , comprising:a fixed beamformer that is adapted to output a first spatially filtered output, and configured to receive a plurality of second spatially filtered outputs from a plurality of steerable beamformers, each steerable beamformer outputting a respective one of the second spatially filtered outputs and associated with a different one of the participants; anda processor capable of determining a similarity between the first spatially filtered output and each of the second spatially filtered outputs, and capable of determining the social interaction between the participants based on the similarity between the first spatially filtered output and each of the second spatially filtered outputs.2. The system of claim 1 , wherein the fixed beamformer comprises a fixed microphone array claim 1 , and each of the steerable beamformers comprises a steerable microphone array.3. The system of claim 1 , wherein the fixed beamformer and the processor are comprised within a mobile device.4. The system of claim 1 , wherein the fixed beamformer and the processor are comprised within one from among the group comprising a ...

SYSTEM AND METHOD FOR WIND DETECTION AND SUPPRESSION

In one embodiment, a pickup system includes a wind detector and a wind suppressor. The wind detector has a plurality of analyzers each configured to analyze first and second input signals, and a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity. The analyzers can be selected from a group of analyzers including a spectral slope analyzer, a ratio analyzer, a coherence analyzer, a phase variance analyzer and the like. The wind suppressor has a ratio calculator configured to generate a ratio of the first and second input signals, and a mixer configured to select one of the first or second input signals and to apply to the selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio. 163-. (canceled)64. A pickup system , the pickup system comprising:a wind detector configured to receive first and second input signals, the wind detector including:a plurality of analyzers each configured to analyze the first and second input signals; anda combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity; anda wind suppressor including:a ratio calculator configured to generate a ratio of sub-band powers of the first and second input signals; anda mixer configured to select one of the first or second input signals and to apply to said selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio, the other of the first or second input signals being unselected.65. The pickup system of claim 64 , wherein:application of the first or second panning coefficients is a function of a ratio of the first and second input signals; and {'br': None, 'sup': '−2*WindLevel(Ratio−RatioTgt)/20', 'α=10Ratio−RatioTgt>0'}, 'one of the first or second ...

Optimum broadcast audio capturing apparatus, method and system

The OPTIMUM BROADCAST AUDIO CAPTURING APPARATUS, METHOD AND SYSTEM (“OBAC”) transforms selection request, video feeds, and, audio feeds inputs via OBAC components into synchronized and optimum video and audio outputs. In one embodiment, a processor-implemented method for capturing optimal audio in sports broadcasting is disclosed, comprising: receiving real time data from a first instrumented camera situated in a venue; processing the real time data to determine a first field position in the venue; activating a first microphone from a plurality of microphones in a first microphone array based on the first field position; and sending a first audio selection from the first microphone.

SOUND PROCESSOR, SOUND PROCESSING METHOD, AND COMPUTER PROGRAM PRODUCT

According to one embodiment, sound processor includes: communication module; outputting module; recording module; display; input module; controller; and calculating module. The controller (i) displays, on display, message prompting user to move a sound input device to position proximate to speaker, (ii) causes the outputting module to output the test sound and causes the recording module to record first sound, (iii) displays, after the first sound is recorded, on the display, message prompting the user to move the sound input device to listening position, and (iv) causes the outputting module to output the test sound and causes the recording module to record second sound. The calculating module finds a first frequency characteristic of the first sound and a second frequency characteristic of the second sound, and calculates, based on a difference between the first and second frequency characteristics, a value for correcting the second frequency characteristic to a target frequency characteristic. 1. A sound processor comprising:a communication module configured to communicate with a sound device;a test sound outputting module configured to cause the sound device to output test sound through the communication module;a recording module configured to record sound collected with a sound input device;a display configured to display a message;an input module configured to receive a user input;a controller configured to (i) display, on the display, a first message prompting a user to move the sound input device to a position proximate to a speaker of the sound device so as to record first sound, (ii) cause the test sound outputting module to output the test sound in accordance with a user input made with respect to the input module in response to the first message and cause the recording module to record the first sound, (iii) display, after the first sound is recorded, on the display, a second message prompting the user to move the sound input device to a listening ...

SYSTEM AND METHOD FOR DATA PROCESSING USING EARPHONE PORT

In order to solve the above-described problem, the present disclosure provides a system and a method for signal processing with an external smart device using an earphone port. The earphone port includes a speaker terminal and a microphone terminal and receives a signal from an earphone plug of the peripheral device connected to the smart device. The connection recognition unit recognizes a connection between the earphone plug of the peripheral device and the earphone port. The application operating unit receives an executive command of an existing signal processing application through an input module and executes the signal processing application in response to the executive command. The data processing unit performs a predetermined signal analysis on a data signal input from the peripheral device through the microphone terminal according to an operation of the executed signal processing application and outputs a result of the signal analysis through an output module. 1. A smart device for performing signal processing with respect to a peripheral device by using an earphone port , the smart device comprising:an earphone port that includes a speaker terminal and a microphone terminal and receives a signal from an earphone plug of the peripheral device connected to the smart device;a connection recognition unit that recognizes a connection between the earphone plug of the peripheral device and the earphone port;an application operating unit that receives an executive command of an existing signal processing application through an input module and executes the signal processing application in response to the executive command; anda data processing unit that performs a predetermined signal analysis on a data signal input from the peripheral device through the microphone terminal according to an operation of the executed signal processing application and outputs a result of the signal analysis through an output module.2. The smart device of claim 1 , further comprising: ...

MICROPHONE OCCLUSION DETECTOR

Digital signal processing microphone occlusion detection is described that can be used with a noise suppression system that uses two types of noise estimators, including a more aggressive one based on two audio signals (such as for non-stationary noises) and a less aggressive one based on one audio signal (such as for stationary noises). Decisions are made on how to select or combine the outputs of the noise estimators into a usable noise estimate, based on an occlusion function. The occlusion detection may alternatively be used to trigger an alert to users of multi-microphone audio processing systems, such as smart phones, headsets, laptops and tablet computers. Other embodiments are also described and claimed. 1. An electronic system for audio noise processing and for noise reduction , using a plurality of microphones , comprising:a first noise estimator to process a first audio signal from a first one of the microphones, and generate a first noise estimate;a second noise estimator to process the first audio signal, and a second audio signal from a second one of the microphones, in parallel with the first noise estimator, and generate a second noise estimate;a combiner-selector to receive the first and second noise estimates, and to generate an output noise estimate using the first and second noise estimates; anda microphone occlusion detector to process the first and second audio signals to generate a microphone occlusion signal that represents a measure of how much one of the microphones is occluded, andwherein the combiner-selector is to generate its output noise estimate also based on the occlusion signal.2. The system of wherein the combiner-selector selects the first noise estimate for its output noise estimate claim 1 , and not the second noise estimate claim 1 , when the occlusion signal indicates that the second one of the microphones is substantially occluded.3. The system of wherein the occlusion detector computes a power or energy ratio (PR) or a ...

SYSTEMS AND METHODS FOR DETERMINING THE CONDITION OF MULTIPLE MICROPHONES

Systems and methods for determining the operating condition of multiple microphones of an electronic device are disclosed. A system can include a plurality of microphones operative to receive signals, a microphone condition detector, and a plurality of microphone condition determination sources. The microphone condition detector can determine a condition for each of the plurality of microphones by using the received signals and accessing at least one microphone condition determination source. 1. A method for determining the operating conditions of microphones of an electronic device , the method comprising:receiving signals from a plurality of microphones;providing at least one microphone condition determination source;providing the signals to a microphone condition detector; andaccessing, using the microphone condition detector, at least one of the at least one microphone condition determination source in conjunction with the signals to determine an operating condition for each of the plurality of microphones.2. The method of claim 1 , wherein the accessing comprises receiving information from the at least one of the at least one microphone condition determination source.3. The method of claim 1 , wherein the operating condition comprises one of a free-field and an interference state.4. The method of further comprisingreceiving device centric data; andusing the device centric data to determine a likelihood of microphone occlusion.5. The method of claim 4 , wherein the device centric data comprises orientation data of the device.6. The method of claim 4 , wherein the device centric data comprises at least one of ambient light data and proximity data.7. The method of claim 1 , wherein the plurality of microphones comprises three or more microphones located on different planes of the device.8. The method of claim 1 , wherein the at least one microphone condition determination source comprises at least one of an a priori database claim 1 , a pattern recognizer claim 1 ...

MEASURING APPARATUS AND COMPUTER PROGRAM

According to one embodiment, a measuring apparatus includes: an output module configured to temporally exclusively output, to a measuring target system, a first output signal corresponding to a first measuring signal for sweeping frequency and a second output signal corresponding to a second measuring signal for sweeping frequency and having a different amplitude characteristic from an amplitude characteristic of the first measuring signal; and a frequency characteristic computation module configured to synthesize a first frequency amplitude spectrum obtained from a first reception signal when a sound output from the measuring target system based on the first output signal is received and a second frequency amplitude spectrum obtained from a second reception signal when a sound output from the measuring target system based on the second output signal is received to generate frequency characteristic data representing an acoustic characteristic. 1. A measuring apparatus comprising:an output module configured to temporally exclusively output, to a measuring target system, a first output signal corresponding to a first measuring signal for sweeping frequency and a second output signal corresponding to a second measuring signal for sweeping frequency and having a different amplitude characteristic from an amplitude characteristic of the first measuring signal; anda frequency characteristic computation module configured to synthesize a first frequency amplitude spectrum obtained from a first reception signal when a sound output from the measuring target system based on the first output signal is received and a second frequency amplitude spectrum obtained from a second reception signal when a sound output from the measuring target system based on the second output signal is received to generate frequency characteristic data representing an acoustic characteristic.2. The measuring apparatus of claim 1 , whereinthe output module is configured to use, as the second measuring ...

Sensor Fusion to Improve Speech/Audio Processing in a Mobile Device

The disclosed system and method for a mobile device combines information derived from onboard sensors with conventional signal processing information derived from a speech or audio signal to assist in noise and echo cancellation. In some implementations, an Angle and Distance Processing (ADP) module is employed on a mobile device and configured to provide runtime angle and distance information to an adaptive beamformer for canceling noise signals, provides a means for building a table of filter coefficients for adaptive filters used in echo cancellation, provides faster and more accurate Automatic Gain Control (AGC), provides delay information for a classifier in a Voice Activity Detector (VAD), provides a means for automatic switching between a speakerphone and handset mode of the mobile device, or primary microphone and reference microphones and assists in separating echo path changes from double talk. 1. A computer-implemented method performed by one or more processors of a mobile device , comprising:receiving data from one or more sensors of a mobile device;calculating an orientation and distance of a signal source relative to a first microphone of the mobile device based on the data;receiving a signal from the source through the first microphone; andprocessing the signal based on the calculated orientation and distance.2. The method of claim 1 , where processing comprises:calculating a gain based on the distance; andautomatically applying the gain to the signal received through the first microphone.3. The method of claim 2 , where automatically applying the gain claim 2 , comprises:comparing the calculated gain with an estimated gain; anddetermining whether to apply the calculated gain to the signal based on results of the comparison.4. The method of claim 3 , where processing comprises:determining a gain error based on the calculated gain and the estimated gain; andapplying either the calculated gain or the estimated gain to the signal received through the ...

GROUPING SYSTEM

Waveform generating section generates a waveform, and waveform collecting sections - to - collect the waveform generated by waveform generating section . Waveform comparing section compares waveforms collected by waveform collecting sections - to - with each other. Group generating section groups waveform collecting sections - to - which collected identical waveforms, into one group as a result of comparison. 1. A grouping system , comprising:a waveform generating section that generates a waveform;waveform collecting sections that collect the waveform generated by said waveform generating section;a waveform comparing section that compares waveforms collected by said waveform collecting sections with each other; anda group generating section that groups waveform collecting sections that collect identical waveforms into one group as a result of comparison by said waveform comparing section.2. The grouping system according to claim 1 , whereinsaid waveform comparing section compares feature values of the waveforms with each other, andsaid group generating section groups waveform collecting sections that collect waveforms identical in feature value into one group as a result of comparison by said waveform comparing section.3. The grouping system according to claim 1 , comprisinga waveform control section that controls generation of the waveform in said waveform generating section, whereinsaid waveform control section reports waveform information indicating the waveform generated by said waveform generating section to said waveform comparing section, andsaid waveform comparing section performs the comparison by using the waveform indicated by the waveform information reported from said waveform control section.4. The grouping system according to claim 1 , whereinsaid waveform collecting section outputs to said waveform comparing section timing information indicating timing during which the waveform was collected together with waveform information indicating the collected ...

CIRCUIT FOR TESTING BUZZER

A circuit for testing a buzzer includes a microphone, an amplifier circuit, a microprocessor circuit, and a display circuit. The amplifier circuit is configured to amplify an analog signal from the microphone to an amplified signal. The microprocessor circuit determines whether a loudness and a frequency of the amplified signal are within proper ranges. The buzzer is qualified if the loudness and frequency of the amplified signal are within the proper ranges. The display unit displays the loudness, the frequency, and a test result. 1. A circuit for testing a buzzer , comprising:a microphone configured to generate an analog signal corresponding to a sound from the buzzer, by playing an audio signal corresponding to a file;an amplifier circuit configured to amplify the analog signal from the microphone, and output an amplified signal;a microprocessor circuit performing an analog to digital (A/D) conversion operation on the amplified signal, to obtain a first loudness of the amplified signal, and sampling the amplified signal to obtain a first frequency of the amplified signal; the microprocessor unit circuit determining whether the first loudness is within a first predetermined range, and determining whether the first frequency is within a second predetermined range; anda display unit configured to display the first loudness, the first frequency, and a test result of the buzzer.2. The circuit of claim 1 , wherein the microprocessor circuit further determines whether the amplified signal has a high frequency.3. The circuit of claim 2 , further comprising a gain circuit claim 2 , wherein the gain circuit is coupled to the amplifier circuit claim 2 , to receive the amplified signal claim 2 , and output a re-amplified signal in response that the amplified signal has a low frequency determined by the microprocessor circuit claim 2 , the microprocessor circuit further determines whether a second loudness corresponding to the re-amplified signal is within the first ...

Detection of Device Configuration

Devices connected with a computer system are interrogated to detect whether they are configured and functioning. In an audio device implementation, a known audio sample is output to a loudspeaker. If the loudspeaker is properly configured and functional, an attached microphone will capture the sound corresponding to the audio sample generated by the loudspeaker. If sound corresponding to the audio sample is detected, an indication is provided to the user that the microphone and loudspeaker are operational. If no sound corresponding to the audio sample is detected, an indication is provided to the user that the loudspeaker is not enabled and additional configuration is required. 1. A method for detection of configuration of a device connected with a computer system , the method comprising:transmitting a sample signal for output via an output device connected with the computer system;capturing a captured signal via an input device connected with the computer system; andcomparing the sample signal to the captured signal to determine whether the captured signal substantially corresponds to the sample signal thus indicating whether the output device, the input device, or both are functional.2. The method of further comprising outputting the sample signal via the output device.3. The method of claim 1 , wherein the step of comparing further comprises processing at least one of the sample signal and the captured signal to transform the sample signal claim 1 , the captured signal claim 1 , or both into a common format.4. The method of claim 1 , wherein the output device comprises a loudspeaker and the input device comprises a microphone.5. The method of claim 1 , wherein the output device comprises a video display device and the input device comprises a video input device.6. The method of claim 1 , wherein the output device comprises a graphic output device and the input device comprises an optical input device.7. The method of further comprising providing an indication to ...

Multifunction System and Method for Integrated Hearing and Communication with Noise Cancellation and Feedback Management

Systems, devices and methods for communication include an ear canal microphone configured for placement in the ear canal to detect high frequency sound localization cues. An external microphone positioned away from the ear canal can detect low frequency sound, such that feedback can be substantially reduced. The canal microphone and the external microphone are coupled to a transducer, such that the user perceives sound from the external microphone and the canal microphone with high frequency localization cues and decreased feedback. Wireless circuitry can be configured to connect to many devices with a wireless protocol, such that the user can receive and transmit audio signals. A bone conduction sensor can detect near-end speech of the user for transmission with the wireless circuitry in noisy environment. Noise cancellation of background sounds near the user can improve the user's hearing of desired sounds.