СПОСОБ ЭЛЕКТРОННОГО ФОРМИРОВАНИЯ ДИАГРАММЫ НАПРАВЛЕННОСТИ ДЛЯ АКУСТИЧЕСКИХ СИГНАЛОВ И УСТРОЙСТВО АКУСТИЧЕСКОГО ДАТЧИКА

Изобретение относится к акустике, конкретно к методу формирования луча по акустическим сигналам. Предварительно определенная характеристика усиления между электрическим приемным сигналом Sr и акустическими сигналами в зависимости от направления θ, с которого поступают указанные акустические сигналы в два разнесенных по пространству акустоэлектрических преобразователя 1,2, формируется таким образом, что повторный взаимно задержанный сигнал А10 определяется из выходных сигналов преобразователей и в соответствии с принятой задержкой преобразователя. При этом один S1 из выходных сигналов фильтруется таким образом, что переходной характеристикой фильтрации управляют в зависимости от взаимно задержанного сигнала A12. Выходной сигнал фильтрации 14 используется в качестве электрического принимаемого сигнала Sr. В результате устройство, выполненное с учетом данных преобразований, обладает хорошей частотной характеристикой, может быть выполнено небольших размеров, а также с высоким отношением сигнал ...

СПОСОБ, УСТРОЙСТВО И ИНСТРУМЕНТ ЭЛЕКТРОННОЙ ПОДПИСИ ДЛЯ ИСПОЛЬЗОВАНИЯ В СОГЛАСОВАНИИ С АУДИОИНТЕРФЕЙСОМ МОБИЛЬНОГО ТЕРМИНАЛА

Изобретение относится к области электронной техники. Технический результат - обеспечение согласования мобильного терминала с аудиоинтерфейсом. Для этого способ предусматривает соединение третьего вывода аудиоинтерфейса с первым выводом через первый контур, который содержит первый модуль односторонней проводимости, с направлением тока к первому выводу; соединение третьего вывода со вторым выводом аудиоинтерфейса через второй контур, содержащий второй модуль односторонней проводимости, с направлением тока ко второму выводу; включение первого модуля односторонней проводимости таким образом, что третий вывод и первый вывод образуют первый замкнутый контур, имеющий импеданс, согласующийся с идентификационным сопротивлением мобильного терминала; и включение второго модуля односторонней проводимости таким образом, что третий вывод и второй вывод образуют второй замкнутый контур, имеющий импеданс, согласующийся с идентификационным сопротивлением мобильного терминала. Это позволяет импедансу замкнутой ...

ДЕТЕКТОРНОЕ УСТРОЙСТВО

Изобретение обеспечивает такое детекторное устройство, которое было бы в состоянии определять, работает ли подключенный к нему микрофон или прибор обработки аудиосигнала в аналоговом или цифровом режиме. В соответствии с этим определением выбирается аналоговый или цифровой разъем. Предусмотрено детекторное устройство, которое имеет устройство (6) сбора данных, устройство (5) распознавания и устройство (4) выборки. Устройство (6) сбора данных обнаруживает внешний сигнал, поданный детекторным устройством и передает этот сигнал на устройство (5) распознавания. Устройство (5) распознавания определяет, был ли зарегистрированный внешний сигнал передан с аналогового или цифрового устройства обработки аудиосигнала и выдает соответствующий определяющий сигнал на устройство (4) выборки. Устройство (4) выборки затем выбирает либо аналоговый, либо цифровой разъем в зависимости от определяющего сигнала. 3 н. и 13 з.п. ф-лы, 2 ил.

УСТРОЙСТВО ДЛЯ ОБРАБОТКИ ЗВУКОВЫХ СИГНАЛОВ

Изобретение относится к акустике, в частности к средствам обработки и передачи звуковой и речевой информации. Способ обработки звуковых сигналов заключается в определении изменения положения устройства, обработке звукового сигнала для передачи по восходящей линии связи в зависимости от изменения положения. При этом звуковой сигнал включает по меньшей мере один звуковой сигнал, полученный от по меньшей мере одного микрофона, и обработку звукового сигнала в зависимости от указанного изменения положения, а также выбор по меньшей мере одного указанного звукового сигнала для вывода в зависимости от изменения положения. Устройство для обработки звуковых сигналов содержит процессор, схему памяти, содержащую компьютерный программный код, схему интерфейса, схему датчиков, микрофонную решетку, аналого-цифровой и цифроаналоговый преобразователи, схему приемо-передатчика. Технический результат - улучшение характеристики сигнал/шум. 2 н. и 14 з.п. ф-лы, 9 ил.

РЕГУЛИРОВКА ГРОМКОСТИ НА ОСНОВАНИИ МЕСТОПОЛОЖЕНИЯ СЛУШАТЕЛЯ

Изобретение относится к средствам регулировки громкости на основании местоположения слушателя. Технический результат заключается в осуществлении возможности регулирования громкости на основании местоположения слушателя. Идентифицируется местоположение одного или более громкоговорителей и отслеживается местоположение слушателя. Для каждого из одного или более громкоговорителей оценивается изменяющееся расстояние между данным громкоговорителем и слушателем. Громкость данного громкоговорителя автоматически регулируется в реальном времени на основании текущего расстояния между данным громкоговорителем и слушателем. 2 н. и 13 з.п. ф-лы, 10 ил.

СИСТЕМЫ И СПОСОБЫ ДЛЯ НАСТРОЙКИ ЭЛЕКТРОННОГО УЛУЧШЕНИЯ ЗВУКА

Раскрыты примерные системы и способы для настройки электронного улучшения звука. Примерный раскрытый способ включает в себя осуществление дискретизации с помощью микрофона, профиля звукового сигнала шумов двигателя в кабине транспортного средства. Примерный способ также включает в себя сравнение дискретизированного профиля звукового сигнала с модельным профилем звукового сигнала. Дополнительно, примерные способы включают в себя,когда дискретизированный профиль звукового сигнала не удовлетворяет пороговому значению подобия, применение функции сближения к дискретизированному профилю звукового сигнала для формирования целевого профиля звукового сигнала и выполнение итераций до тех пор, пока целевой профиль звукового сигнала не удовлетворяет пороговому значению подобия. 3 н. и 14 з.п. ф-лы, 5 ил.

УСТРОЙСТВО, СОДЕРЖАЩЕЕ МНОЖЕСТВО АУДИОДАТЧИКОВ, И СПОСОБ ЕГО ЭКСПЛУАТАЦИИ

Изобретение относится к акустике, в частности к средствам обработки звуковой информации, полученной от микрофонов костной и воздушной проводимости. Устройство содержит множество аудиодатчиков, первый из которых контактирует с пользователем устройства, а второй контактирует с воздухом. Осуществляют прием аудиосигналов, анализируя которые, различают тип аудиопреобразователя, находящегося в контакте с пользователем. При этом анализ включает в себя преобразование Фурье, нормализацию преобразованных сигналов с последующей оценкой мощности аудиосигнала и её сравнением с пороговой величиной. Затем воспроизводят выходной аудиосигнал, представляющий речь пользователя. Устройство также содержит дискриминатор, соединенный входами с выходами микрофонов, детектор речи, схему улучшения речи, схему выделения признаков, схему коррекции аудиосигналов, передающую схему. 3 н. и 10 з.п. ф-лы, 14 ил.

СПОСОБ ФОРМИРОВАНИЯ ХАРАКТЕРИСТИК КОНДЕНСАТОРНОГО МИКРОФОНА И УСТРОЙСТВО ДЛЯ ЕГО ОСУЩЕСТВЛЕНИЯ

Изобретение относится к электроакустика и может быть использовано в аппаратуре обработки звука, в различных областях промышленности, где в аппаратуре используется датчик - электростатический преобразователь-генератор. Техническим результатом данного способа и устройства является повышение качества работы конденсаторного микрофона за счет снижения искажений звука и получения стабильных характеристик микрофона. В способе формирования характеристик конденсаторного микрофона, включающем подачу сигнала по крайней мере с одного неподвижного (подвижного) электрода электростатического преобразователя-генератора на усилитель, обработку данного сигнала усилителем и подачу его части на звукообрабатывающее устройство, часть сигнала с выхода усилителя подают по крайней мере на один подвижный (неподвижный) электрод электростатического преобразователя-генератора посредством обратной связи, которой дополнительно охватывают электростатический преобразователь-генератор. Устройство, содержащее электростатический ...

СПОСОБ ПОЛУЧЕНИЯ ЗВУКОВОЙ ИНФОРМАЦИИ

Изобретение относится к области звуковой техники, а именно к способам получения звуковой информации от биологических и других объектов. Предложен способ получения звуковой информации, заключающийся в приеме звуковых волн через воздушное пространство, при этом дополнительно принимают звуковые волны путем непосредственного контакта с источником звука и суммируют принятые сигналы в устройстве пропорционального суммирования. В результате повышается достоверность и придаются новые качества результирующему звуковому сигналу при приеме звуковых волн от различных объектов. 1 ил.

СПОСОБ КОРРЕКТИРОВКИ НАПРАВЛЕНИЯ ОСИ ОТРАЖАТЕЛЬНОГО ПРИЕМНИКА ЗВУКОВЫХ ВОЛН НА ВИЗУАЛЬНО ТРУДНО НАБЛЮДАЕМЫЕ ИЛИ НЕНАБЛЮДАЕМЫЕ ИСТОЧНИКИ ЗВУКА

Изобретение относится к акустике, в частности к способам приема звука с помощью остронаправленного микрофона. Способ корректировки направления оси отражательного приемника звуковых волн включает в себя операцию приема звука параболическим отражателем, в фокусе которого помещают направленный микрофон. К ободу отражателя диаметром Dжестко крепят штангу, на которой устанавливают симметрично два дополнительных микрофона, расстояние между которыми изменяют в соответствии с неравенством LF>F. Выходной сигнал суммирующего усилителя через пороговое устройство подают на регистратор, на котором фиксируют максимальное значение сигнала, включают на прием звука микрофон, размещенный в фокусе отражателя, а выходной сигнал с этого микрофона подключают через фильтр с полосой пропускания 60 Гц к усилителю, выходной сигнал которого подают на ...

СПОСОБ И УСТРОЙСТВО ДЛЯ ЗАХВАТА АУДИОИНФОРМАЦИИ С ИСПОЛЬЗОВАНИЕМ ФОРМИРОВАНИЯ ДИАГРАММЫ НАПРАВЛЕННОСТИ

Изобретение относится к акустике. Устройство для приема аудиоинформации содержит: массив микрофонов, один формирователь диаграммы направленности, множество ограниченных формирователей диаграммы направленности, первый адаптер для адаптации параметров диаграммы направленности первого формирователя диаграммы направленности, второй адаптер для адаптации параметров ограниченной диаграммы направленности для множества ограниченных формирователей диаграммы направленности, процессор разности для вычисления разности между ограниченными диаграммами направленности. Второй адаптер выполнен с возможностью адаптировать параметры ограниченной диаграммы направленности с тем ограничением, что параметры ограниченной диаграммы направленности адаптируются только для тех ограниченных формирователей диаграммы направленности из множества ограниченных формирователей диаграммы направленности, для которых была определена мера различия, соответствующая критерию сходства, и причем процессор разности выполнен с возможностью ...

ПРЕОБРАЗОВАТЕЛЬ ПЕРЕМЕННОГО ТОКА В ПОСТОЯННЫЙ

... 1. Преобразователь переменного тока в постоянный, содержащий силовые трансформаторы, к первичным обмоткам которых подключены управляемые ключевые элементы, а к вторичным обмоткам - неуправляемые вентили по шестифазной нулевой схеме выпрямления, отличающийся тем, что в него дополнительно введены двухполюсные элементы, подключенные между вторичными обмотками трансформаторов.2. Преобразователь по п.1, отличающийся тем, что двухполюсные элементы выполнены в виде активных сопротивлений.

ДИНАМИЧЕСКИЙ ГРОМКОГОВОРИТЕЛЬ

... 1. Динамический громкоговоритель, содержащий магнитную систему, которая включает в себя цилиндрический корпус из магнитомягкого железа, дисковый концентратор магнитного потока в виде торцевой стенки цилиндрического корпуса, как минимум два кольцевых постоянных магнита с осевой намагниченностью, закрепленные с помощью тонкостенного крепежного стакана из диамагнетика внутри цилиндрического корпуса соосно с ним на торцевой стенке, керн из магнитомягкого железа, кольцевой концентратор магнитного потока из магнитомягкого железа, первый каркас с внешней кольцевой катушкой и второй каркас с внутренней кольцевой катушкой, которые закреплены соосно на общей дисковой связке из немагнитного материала, связанной центрирующей шайбой и диффузором, отличающийся тем, что кольцевые постоянные магниты с осевой намагниченностью размещены на тонкостенном крепежном стакане с зазором между встречно расположенными одноименными полюсами, внешняя кольцевая катушка, закрепленная на первом каркасе, размещена внутри ...

Последовательный СЧ-, ВЧ-фильтр для автомобильных акустических систем

Полезная модель относится к радиотехнике и может быть использована в акустических системах бытовой или профессиональной звуковоспроизводящей аппаратуры. Последовательный СЧ-, ВЧ-фильтр для автомобильных акустических систем содержит частотный фильтр средних частот и частотный фильтр верхних частот, выходы которых предназначены для подключения динамических головок соответственно среднечастотного и высокочастотного диапазона, при этом в направлении от плюса к минусу последовательно соединены между собой выводами первый конденсатор, катушка индуктивности и второй конденсатор, причем выводы динамической головки среднечастотного диапазона соединены: один вывод - с общим выводом второго конденсатора и катушки индуктивности, а другой вывод - с подключенным к минусу вторым выводом второго конденсатора, выводы динамической головки высокочастотного диапазона соединены: один вывод через третий конденсатор с общим выводом катушки индуктивности и второго конденсатора, а другой вывод через четвертый конденсатор ...

Микрофонный усилитель

Изобретение относится к акустике. Микрофонный усилитель содержит высокоомный резистор, фильтр поляризующего напряжения, переключаемый делитель напряжения, умножитель напряжения, входной разделительный конденсатор, повторитель, инвертор, выходные разделительные конденсаторы, генератор переменного напряжения, линейный стабилизатор напряжения, фильтр входного питания, резисторы подачи питающего напряжения, ЭМС фильтр. Входной повторитель выполнен на основе операционного усилителя с высокоомным входом, сигнал на который подаётся с микрофонного капсюля через разделительный конденсатор, а поляризующее напряжение на микрофонный капсюль подаётся через промежуточные элементы с умножителя напряжения, на который поступает переменное напряжение с генератора, от линейного стабилизатора запитаны операционные усилители и генератор, с выхода повторителя сигнал подаётся на один выход усилителя и на вход инвертора, выполненного на операционном усилителе, с выхода которого сигнал поступает на другой выход ...

ДИНАМИЧЕСКИЙ ГРОМКОГОВОРИТЕЛЬ

... 1. Динамический громкоговоритель, содержащий магнитную систему, которая включает в себя цилиндрический корпус из магнитомягкого железа, торцевую стенку из магнитомягкого железа, как минимум четыре кольцевых постоянных магнита с осевой намагниченностью, закрепленные с помощью тонкостенных крепежных стаканов из диамагнетика внутри цилиндрического корпуса соосно с ним на торцевой стенке, керн из магнитомягкого железа, кольцевые концентраторы магнитного потока из магнитомягкого железа, первый каркас с внешней кольцевой катушкой, второй каркас с промежуточной кольцевой катушкой и третий каркас с внутренней кольцевой катушкой, которые закреплены соосно на общей дисковой связке из немагнитного материала, связанной с центрирующей шайбой и диффузором, отличающийся тем, что кольцевые постоянные магниты размещены в разных параллельных плоскостях по два кольцевых постоянных магнита в каждой плоскости, причем первые из которых - внутренние кольцевые постоянные магниты, имеющие внешний диаметр меньше ...

ПРЕОБРАЗОВАТЕЛЬ НАПРЯЖЕНИЯ ДЛЯ КАТОДНОЙ ЗАЩИТЫ

Преобразователь напряжения предназначен для катодной защиты от коррозии металлоконструкций в химической и нефтегазовой промышленности. Преобразователь напряжения содержит сетевой фильтр радиопомех, подключенный к сетевому выпрямителю, активный фильтр, нагрузкой которого является высокочастотный инвертор, преобразующий постоянное напряжение в высокочастотное переменное для питания трансформатора. Вторичная обмотка трансформатора соединена с выпрямительным мостом. Преобразователь напряжения имеет также выходной сглаживающий фильтр для сглаживания пульсаций выходного напряжения, подключенный к выпрямительному мосту. Новым в заявляемом преобразователе является то, что он снабжен управляемым выходным напряжением дросселем высокочастотного инвертора, включенным последовательно с первичной обмоткой трансформатора. Это повысило надежность и устойчивость работы преобразователя.

ДИНАМИЧЕСКИЙ ГРОМКОГОВОРИТЕЛЬ

... 1. Динамический громкоговоритель, содержащий магнитную систему, которая включает в себя цилиндрический корпус из магнитомякого железа, торцевую стенку из магнитомягкого железа, керн из магнитомягкого железа, как минимум четыре кольцевых постоянных магнита с осевой намагниченностью, закреплённых с помощью тонкостенных крепёжных стаканов из диамагнетика внутри цилиндрического корпуса соосно с ним на торцевой стенке, кольцевые концентраторы магнитного потока из магнитомягкого железа, первый каркас с внешней кольцевой катушкой, второй каркас с средней кольцевой катушкой и третий каркас с внутренней кольцевой катушкой, из которых первый каркас с внешней кольцевой катушкой размещён в зазоре между внутренней поверхностью цилиндрического корпуса и внешней образующей внешних кольцевых постоянных магнитов, второй каркас с средней кольцевой катушкой размещён в зазоре между внутренней образующей внутренних кольцевых постоянных магнитов и внутренней поверхностью первого тонкостенного крепёжного стакана ...

УСТРОЙСТВО И СПОСОБ УДАЛЕНИЯ ВИБРАЦИИ ДЛЯ НАУШНИКОВ С ДВУМЯ МИКРОФОНАМИ

Изобретение относится к наушникам. Наушники содержат динамик, микрофон и вибрационный датчик. Микрофон выполнен с возможностью приёма первого сигнала, включающего в себя голосовой сигнал, и первого вибрационного сигнала. Вибрационный датчик выполнен с возможностью приёма второго вибрационного сигнала, причём первый вибрационный сигнал и второй вибрационный сигнал происходят от вибрации источника вибрации; и объем полости вибрационного датчика больше, чем объём полости микрофона таким образом, что микрофон и вибрационный датчик имеют приблизительно одинаковую частотную характеристику по отношению к вибрации источника вибрации. Технический результат – повышение качества звука. 13 з.п. ф-лы, 30 ил., 1 табл.

СПОСОБ ФОРМИРОВАНИЯ ВОЗБУЖДАЮЩЕГО НАПРЯЖЕНИЯ ПЬЕЗОЭЛЕКТРИЧЕСКОГО ПРЕОБРАЗОВАТЕЛЯ И УСТРОЙСТВО ДЛЯ ЕГО ОСУЩЕСТВЛЕНИЯ

Использование: измерительная техника, приборы, ультразвуковые датчики. Сущность изобретения: формируют возбуждающее напряжение пьезоэлектрического преобразователя, выделяя напряжение пьезоэффекта и усиливая его по мощности до необходимой величины. Устройство содержит усилитель, конденсатор и пять резисторов. Приведено соотношение параметров элементов устройства. 2 с.п. ф-лы, 3 ил.

УСТРОЙСТВО ДЛЯ ЭЛЕКТРОМЕХАНИЧЕСКОГО ПРЕОБРАЗОВАНИЯ ЗВУКОВОЙ ЧАСТОТЫ

Применение: устройство предназначено для уменьшения искажений электромеханического преобразования и может использоваться при производстве высококачественных звуковоспроизводящих устройств. Сущность изобретения: устройство содержит усилитель с основным и дополнительным входами, звуковую катушку динамической головки, электромеханический трансформатор, состоящий из передающей и приемной катушек и цепь коррекции. 1 ил.

ЭЛЕКТРОННОЕ УСТРОЙСТВО

Изобретение относится в акустике, в частности к устройствам для передачи акустического сигнала к пользователю. Устройство содержит пьезоэлектрический элемент, вибрационную пластину, удерживаемую корпусом, причем вибрационная пластина выполнена с возможностью колебания пьезоэлектрическим элементом, а звуковые колебания, передаваемые вибрационной пластиной, воспринимаются ушной раковиной пользователя посредством костной и воздушной проводимости. Пьезоэлектрический элемент колеблется за счет отфильтрованного звукового сигнала, в котором удалена часть компонентов, превышающих заданное пороговое значение. Вибрационная пластина выполнена с возможностью изгибаться пьезоэлектрическим элементом таким образом, что часть вибрационной пластины непосредственно над пьезоэлектрическим элементом выступает относительно соседних участков вибрационной пластины в направлении длинной стороны пьезоэлектрического элемента. При этом коэффициент затухания обработанного звукового сигнала по отношению к воспроизводимому ...

ПОДАВЛЕНИЕ ПЕРЕКРЕСТНЫХ ПОМЕХ В ТЕЛЕСНОМ УГЛЕ ДЛЯРЕШЕТОК ФОРМИРОВАНИЯ ДИАГРАММЫ НАПРАВЛЕННОСТИ

... 1. Система для улучшения выходного сигнала преобразователя, содержащая по меньшей мере один преобразователь, формирователь диаграммы направленности, по меньшей мере один выбранный, фиксированный входной луч, алгоритмический блок, вырабатывающий желательный результирующий выходной луч, имеющий суженную осевую ширину луча, и выходной сигнал, соответствующий желательному результирующему выходному лучу, имеющий суженную ширину луча. 2. Система по п.1, отличающаяся тем, что дополнительно содержит множество выбранных, фиксированных входных лучей, причем алгоритмический блок вырабатывает множество желательных выходных лучей, при этом выходной сигнал соответствует множеству желательных выходных лучей, имеющих желательную ширину луча. 3. Система по п.1, отличающаяся тем, что преобразователем является микрофон, который одновременно принимает множество акустических сигналов, которые могут быть пространственно представлены как диаграммы направленности. 4. Система по п.3, отличающаяся тем, что включает ...

СПОСОБ ФОРМИРОВАНИЯ ХАРАКТЕРИСТИК КОНДЕНСАТОРНОГО МИКРОФОНА И УСТРОЙСТВО ДЛЯ ЕГО ОСУЩЕСТВЛЕНИЯ

... 1. Способ формирования характеристик конденсаторного микрофона, включающий подачу сигнала по крайней мере с одного неподвижного (подвижного) электрода электростатического преобразователя-генератора на усилитель, обработку данного сигнала усилителем и подачу его части на звукообрабатывающее устройство, отличающийся тем, что часть сигнала с выхода усилителя подают по крайней мере на один подвижный (неподвижный) электрод электростатического преобразователя-генератора посредством последовательной обратной связи, которой дополнительно охватывают электростатический преобразователь-генератор. 2. Устройство по п.1, содержащее электростатический преобразователь-генератор с подвижными и неподвижными электродами и усилитель, отличающееся тем, что оно дополнительно содержит блок подачи части сигнала с выхода усилителя на подвижный (неподвижный) электрод электростатического преобразователя-генератора. 3. Устройство по п.2, отличающееся тем, что блок подачи части сигнала выполнен в виде прямого соединения ...

ОБРАБОТКА ЗВУКА В ЗВУКОВЫХ УСЛУГАХ С ЭФФЕКТОМ ПРИСУТСТВИЯ

УСТРОЙСТВО И СПОСОБ ДЛЯ УМЕНЬШЕНИЯ ШУМОВ В ЗВУКОВОМ ВХОДНОМ СИГНАЛЕ

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 ПОЛУЧЕНИЯ ИНФОРМАЦИИ О НАПРАВЛЕНИИ И КОМПЬЮТЕРНЫЙ ПРОГРАММЫЙ ПРОДУКТ

... 1. Устройство (100) для получения информации (101, d(k, n)) о направлении от множества микрофонных сигналов (103до 103, Pдо P) или от множества компонентов (P(k, n)) микрофонного сигнала (103, Ρ), в котором различающиеся эффективные направления обзора микрофонов связаны с микрофонными сигналами (103до 103, Pдо P) или компонентами (P(k, n)), при этом устройство (100) содержит:блок (105) объединения, выполненный с возможностью получения значения амплитуды микрофонного сигнала (P) или компонента (P(k, n)) микрофонного сигнала (Pi), и комбинирования единиц (bдо b) информации о направлении, описывающих эффективные направления обзора микрофонов, при этом единица (b) информации о направлении, описывающая данное эффективное направление обзора микрофона, взвешивается в зависимости от значения амплитуды микрофонного сигнала (P) или компонента (P(k, n)) микрофонного сигнала (P), связанного с данным эффективным направлением обзора микрофона, для получения информации (101, d(k, n)) о направлении;в котором ...

МИКРОФОННАЯ УСИЛИТЕЛЬНАЯ СИСТЕМА

Изобретение относится к технике усиления сигналов звуковых частот и предназначено для усиления акустических сигналов. Сущность изобретения: при работе системы микрофоны одинаково воспринимают фоновые звуковые сигналы, имеющие одинаковый уровень. Два почти одинаковых сигнала с обоих микрофонов поступают на входы дифференциального усилителя и дифференциальный усилитель их взаимно вычитает. Поскольку указанные сигналы почти одинаковы, сигнал от фоновых звуков на выходе усилителя практически отсутствует. При работе с микрофонами голос в большей степени воздействует на микрофон, поскольку отверстия и микрофон находятся вблизи рта говорящего, что приводит к разбалансу сигналов на выходе дифференциального усилителя.

ЗВУКОВОСПРОИЗВОДЯЩАЯ СТЕРЕОСИСТЕМА

Изобретение может быть использовано в качестве звуковоспроизводящей стереосистемы в стационарной и автомобильной аппаратуре. Для повышения качества воспроизведения сигналов во всем диапазоне звуковых частот за счет снижения интермодуляционных и нелинейных искажений система содержит аттенюаторы, подключенные через сумматор и усилитель к низкочастотному излучателю, а акустические широкополосные излучатели через фильтры высоких частот подключены к информационным входам системы, которые соединены непосредственно с одним из аттенюаторов, а с другими - через фильтры высоких частот.

ВОЗБУЖДЕНИЕ МНОГОКАНАЛЬНЫХ ГРОМКОГОВОРИТЕЛЕЙ

... 1. Система возбуждения для генерации сигналов возбуждения для аудиовозбудителей, содержащая:разветвитель (107) для генерации первого сигнала и второго сигнала из входного сигнала, причем первый сигнал содержит компоненты сигнала из первого частотного интервала входного сигнала, а второй сигнал содержит компоненты сигнала из второго частотного интервала входного сигнала, причем первый и второй частотные интервалы имеют переходную частоту, и первый частотный интервал соответствует более низким частотам, чем второй частотный интервал;первую схему (111, 115) возбуждения, связанную с разветвителем (107) и выполненную с возможностью генерации первого сигнала возбуждения для первого аудиовозбудителя (105) из первого сигнала;вторую схему (117, 119) возбуждения, связанную с разветвителем (107) и выполненную с возможностью генерации второго сигнала возбуждения для второго аудиовозбудителя (101) из второго сигнала;средство (125) для определения указания отклонения возбудителя для второго аудиовозбудителя ...

СПОСОБ И СИСТЕМА РЕГУЛИРОВКИ УРОВНЯ ГРОМКОСТИ

... 1. Способ для управления выходным сигналом уровня громкости первого устройства (22), причем упомянутое устройство выполнено с возможностью обмена данными со вторым устройством (16) через сетевое соединение, указанный способ содержит этапы, на которых второе устройство отправляет сообщение (42) первому устройству, которое принимает упомянутое сообщение, осуществляет доступ к сохраненной политике (50) и настраивает упомянутый выходной сигнал в зависимости от сохраненной политики. 2. Способ по п.1, в котором политика (50) содержит критерий, который включает в себя время суток для воздействия на упомянутую настройку. 3. Способ по п.1 или 2, в котором политика (50) содержит критерий, который включает в себя количество принятых сообщений в предопределенном временном интервале для воздействия на упомянутую настройку. 4. Способ по п.1, в котором политика (50) содержит критерий, который включает в себя пороговое значение, с которым текущий выходной уровень громкости сравнивается для воздействия ...

ОЦЕНКА ПОЛОЖЕНИЯ ИСТОЧНИКА ЗВУКА

... 1. Устройство для определения оценки положения для источника звука, причем устройство содержитпервый микрофон (M1), предназначенный для генерирования сигнала первого микрофона;второй микрофон (M2), предназначенный для генерирования сигнала второго микрофона;акустический элемент (203), обеспечивающий акустический эффект для звука из положений источника звука для первого микрофона, причем акустический эффект является асимметричным относительно оси (201) между первым и вторым микрофонами;процессор положения, содержащийпервую схему (305) положения для оценки первого возможного положения и второго возможного положения для источника звука, в ответ на время прихода в первой микрофон и во второй микрофон звука от источника звука, причем первое возможное положение представляет собой положение в первой области с одной стороны оси (201), а второе возможное положение представляет собой положение во второй области с противоположной стороны оси (201);блок (307) оценки, предназначенный для определения ...

СПОСОБ ОБРАБОТКИ АКУСТИЧЕСКИХ СИГНАЛОВ, КОМПЬЮТЕРНАЯ ПРОГРАММА И УСТРОЙСТВО ОБРАБОТКИ АКУСТИЧЕСКИХ СИГНАЛОВ

Звуковоспроизводящее устройство

Изобретение относится к радиотехнике и может использоваться в составе различной звуковоспроизводящей аппаратуры. Цель изобретения - уменьшение искажений акустического сигнала. Звуковоспроизводящее устройство состоит из основного усилителя 1 мощности, к выходу которого подключен громкоговоритель 2 с датчиком 3 сигнала обратной связи. Для уменьшения нелинейных искажений, возникающих в цепи основного усилителя 1 мощности и громкоговорителя 2, введен дополнительный канал звуковоспроизведения . Канал состоит из двухвходового сумматора 6, к одному .из входов которого через фазовращатель 7 присоединен вход усилителя 1, а к другому - датчик 3, выход сумматора через корректирующее звено 8 и дополнительный усилитель 4 соединен с дополнительным громкоговорителем 5. Нелинейные искажения, возникающие на выходе громкоговорителя 2, корректируются звуковым полем громко- говоритеЛя 5 за счет интерференционного звукового поля, возникающего при работе обоих громкоговорителей. 1 ил. s Ё ...

Звуковоспроизводящее устройство

Устройство согласования

Установка для озвучивания

УСТАНОВКА ДЛЯ ОЗВУЧИВАНИЯ , содержащая микрофоны, лштю задержки , суммирующре схемы, выхода которых соединены с соответствующими громкоговорителями, отлнчающа51 с я тем, что, с целью уменьшения нскажений звукового поля, в неё евепена дополнительная суммирующая схема, которой соединены с микрофонами, а выход - с входом линии задержки, при этом выходы линии задержки соединены с первыми входами суммирующих схем, вторые входы. которых соединены с соответствующими микрофонами. (Л 4 Сл 5 ...

Устройство для управления трансляционной радиоточкой

Изобретение относится к радиотехнике. Цель изобретения - расширение функциональных возможностей за счет обеспечения автоматического включения трансляционной радиоточки при появлении сигнала в радиотрансляционной сети. Через выпрямитель 2 и конденсатор 6 при появлении сигнала протекает ток, возбуждающий обмотку реле 5. Реле замыкает контакт 4, подключая динамик 3 к сети. После заряда конденсатора 6 этот ток прекращается. Выключение динамика осуществляется нажатием на кнопку 7, подающую импульс тока в обмотку, что вызывает отключение реле 5. 1 ил.

IMPLANTIERBARE VORRICHTUNG ZUR ERFASSUNG VON BIOSIGNALEN MIT AKUSTISCHER ÜBERTRAGUNG

LAUTSPRECHERSYSTEM MIT KONSTANTEM GEHÄUSE-INNENDRUCK UND WANDLER FÜR DIESES SYSTEM

MULTI-CANAL ADAPTIVE ONES LANGUAGE SIGNAL PROCESSING WITH INTERCARRIER NOISE SUPPRESSOR

LOUDSPEAKER SYSTEM FOR VIRTUAL SOUND SYNTHESIS

DIRECTION RECEIPT SYSTEM

SWITCHING CONFIGURATION TO THE FREQUENCY SELF-REGULATION OF AN ULTRASONIC END TRANSDUCER

SYSTEM AND PROCEDURE FOR SIGNAL PROCESSING

CIRCUITS FOR SUPPRESSING GLITCHES DURING THE SYSTEM POWER-ON OF AN AUDIO DEVICE AND, THE SUCH CIRCUITS USING

AUDIO SYSTEM WITH PRECAUTIONS TO THE FILTER COEFFICIENT COPYING

PHASE-CONTROLLED GROUP OF CLAY/TONE RECEIVERS

DIGITAL PULSE WIDTH MODULATION GENERATOR

PROTECTION OF THE EARS CIRCUIT ALSO IN TWO PROTECTION OF THE EARS SHELLS INTRODUCTION ELECTROMAGNETIC ONE HOERKAPSELN

DEVICE WITH A MICROPHONE AND A HF TRANSMITTER FOR CONNECTION WITH THE MICROPHONE

ACOUSTIC MEASURING SYSTEM FOR THE DETECTION OF SOURCES OF INTOXICATION

PROCEDURE AND DEVICE, IT A FINAL USER MAKING POSSIBLE TO STOP LISTENER PREFERENCES FOR HEAR-OBSTRUCTED AND NON--HEAR-OBSTRUCTED

SPITZENWERT-ABGELEITETE ZEITSTEUER-STIMULATIONSSTRATEGIE FÜR EIN MEHRKANAL-COCHLEAR-IMPLANTAT

PROCEDURE FOR THE REDUCTION OF INTERFERENCE POWERS AND EQUIVALENT ACOUSTIC SYSTEM

LINEAR LOUDSPEAKER ARRANGEMENT AND METHOD FOR THE PLACEMENT OF THE TRANSDUCERS

INTERACTIVE LANGUAGE REMAINDER CLAY/TONE HEARING AID AS WELL AS AUXILIARY EQUIPMENT

LANGUAGE TRANSFORMATION EQUIPMENT AND PROCEDURE

LAUTSPRECHER

TONAUFNAHME-EINRICHTUNG

VORRICHTUNG ZUM WOELBEN VON KREISRINGSEGMENT- FOERMIGEN BLECHZUSCHNITTEN ZU KONISCHEN TEILEN

REINFORCEMENT AND SPECTRAL CURVE ADJUSTMENT WITH THE PROCESSING OF AUDIO SIGNALS

PROCEDURE FOR THE OPTIMIZED FILTERING NOT STATIONARY NOISES, THE OF AN AUDIO DEVICE WITH SEVERAL MICROPHONES CAUGHT, IN PARTICULAR A FREE SPEECH TELEPHONE SYSTEM FOR MOTOR VEHICLES

SYSTEM FOR THE DELIMITATION OF LOUDSPEAKER DEFLECTION

AUDIO AMPLIFIER WITH LOCAL INTERFACE

PROCEDURE AND SYSTEM FOR CALIBRATING MICROPHONES

ARRANGEMENT FOR THE CLAY/TONE ADMISSION AND TONWIEDERGABE OF MEANS OF A MULTIPLICITY OF SENSORS

PROCEDURE AND DEVICE FOR THE MICROPHONE ADJUSTMENT FOR A PORTABLE DIRECTION HEARING AID ASSISTANCE OF THE VOICE OF THE CARRIER

LOUDSPEAKER SYSTEM FOR THE RENDITION OF LOW-FREQUENCY ONES OF ACOUSTIC TONES, WHICH WORKS WITH A CLOSED HOUSING OF SMALL VOLUME

HEADPHONE

CLAY/TONE PHOTOGRAPH MECHANISM

ELECTRO-DYNAMIC LOUDSPEAKER

MICROPHONE PLANT WITH SEVERAL MICROPHONES

OVERLOAD PROTECTION CIRCUIT FOR A LOUDSPEAKER ARRANGEMENT

SYSTEMS FOR THE REDUCTION OF SECONDARY ONES OF HOT PLACES IN PHASE-CONTROLLED FOKUSIERTEN ULTRASONIC SYSTEMS

ARRANGEMENT, CONSISTING OF A WIRELESS MICROPHONE AND A RECEIVER

LANGUAGE SIGNAL PROCESSING FOR THE COMMON ADAPTIVE REDUCTION OF INTERFERENCES AND OF ACOUSTIC ECHOES

STÖRSCHALLUNTERDRÜCKENDER KOPFHÖRER

A noise cancelling headphone is described. The noise cancelling headphone utilizes a low power consuming noise cancellation circuit wherein an audio input signal is directly fed into the headphone without the use of an additional headphone amplifier. The noise cancelling circuit uses a microphone to pick up ambient noise and produces a signal which is equal in amplitude but opposite in polarity to the ambient noise signal. The resultant signal is mixed with the audio input signal and fed into the speakers of the headphone. This method is advantageous because it uses fewer components than conventional noise cancellation circuits and it also consumes less power due to the use of fewer components. The distortion of the audio input signal is also reduced since no amplification is performed to the audio input signal onboard the noise cancellation circuit.

PROCEDURE AND ARRANGEMENT FOR THE SIGNAL COMPARISON TO THE TRANSDUCER PRICE INCREASE AND TRANSDUCER PRICE INCREASE SYSTEM

DEVICE FOR THE IMPROVEMENT OF THE BASS RENDITION WITH LOUDSPEAKER SYSTEMS WITH CLOSED HOUSINGS

ELECTROACOUSTIC EQUIPMENT

WARNING SIGNAL GENERATOR FOR THE SUPPLY OF A PRESSURE CHAMBER LOUDSPEAKER.

OSCILLATOR FOR ULTRASCHALLFLSSSSIGKEITSZERSTAEUBER.

PROCEDURE AND ARRANGEMENT FOR THE ELECTRICAL SUPPLY OF SOUND AND ULTRASONIC TRANSDUCERS.

LAUTSPRECHERSYSTEM MIT KONSTANTEM GEHÄUSE-INNENDRUCK UND WANDLER FÜR DIESES SYSTEM

SIMULATION METHOD AND DEVICE (ELECTRONIC ART HEAD) TO THE REPRODUCTION OF THE TRANSMISSION CHARACTERISTICS OF THE HUMAN OUTER EAR DURING FREE FIELD ACOUSTIC IRRADIATION.

ULTRASCHALL-SENDER.

LOUDSPEAKER

GENERATOR FOR FLEXIBLE IMPULSES OF GIVEN DESIRED FORM AND ITS APPLICATION IN MEDICAL TREATMENT OR DIAGNOSIS.

CONFERENCE FREE SPEECH MECHANISM.

AUDIO RENDITION MECHANISM WITH FREQUENCY-SELECTIVE AMPLIFIERS FOR THE CONTROL OF INTERNAL EQUIPMENT ONES AND EXTERNAL LOUDSPEAKERS.

OUTPUT LIMITER FOR A CLASS D BICMOS HEARING AID OUTPUT AMPLIFIER

HOERGERAET WITH A BEHIND THE EAR BASIC HOUSINGS.

ELECTROMECHANICAL FILM AND PROCEDURE FOR ITS PRODUCTION.

SPEECH RECOGNITION SYSTEM.

ION-EMITTING ELECTROACOUSTIC TRANSDUCER.

PROCEDURE AND DEVICE FOR THE ELIMINATION LOUDSPEAKER INTERFERENCE FROM MICROPHONE SIGNALS

PROCEDURE AND SYSTEM FOR THE MEASUREMENT OF THE TRANSMISSION QUALITY OF A SYSTEM

PROCEDURE AND SYSTEM FOR REPRESENTING A SOUND FIELD

Устройство для вывода аудио-сигнала повышенного качества

Полезная модель относится к акустике, в частности к аудио-оборудованию, выполненному с возможностью воспроизводить звук от различных аудио-источников. Устройство содержит процессор, дельта-сигма модулятор, блок усреднения сигнала с интегрированным фильтром сигнала, два операционных усилителя, коммутационный переключатель, интерфейс цифрового линейного ввода, преобразователь импеданса, блок компенсации смещения цифрового звукового сигнала и блок подачи опорного напряжения. При этом блок подачи опорного напряжения соединен с коммутационным переключателем, первый операционный усилитель и интерфейс цифрового линейного ввода подключены к первому входу коммутационного переключателя, а второй операционный усилитель и блок подачи опорного напряжения подключены ко второму входу коммутационного переключателя. Технический результат - повышение качества звука. 1 ил. РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 167 902 U1 (51) МПК H04R 3/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ТИТУЛЬНЫЙ ЛИСТ ОПИСАНИЯ ПОЛЕЗНОЙ МОДЕЛИ К ПАТЕНТУ (21)(22) Заявка: 2016139876, 11.10.2016 (24) Дата начала отсчета срока действия патента: 11.10.2016 (72) Автор(ы): Куницын Игорь Алексеевич (RU) Дата регистрации: 11.01.2017 Приоритет(ы): (22) Дата подачи заявки: 11.10.2016 R U (73) Патентообладатель(и): Общество с ограниченной ответственностью "Музыкальное издательство "Рэй Рекордс" (RU) (45) Опубликовано: 11.01.2017 Бюл. № 2 1 6 7 9 0 2 R U (57) Формула полезной модели Устройство для вывода аудио-сигнала, включающее процессор и аудиовыход, отличающееся тем, что дополнительно содержит: последовательно соединенный с процессором дельта-сигма модулятор; блок усреднения сигнала с интегрированным фильтром сигнала, соединенный с дельта-сигма модулятором; первый операционный усилитель A1 и второй операционный усилитель A2, параллельно подключенные к блоку усреднения сигнала; коммутационный переключатель, подключенный к выходам первого и второго усилителей и процессора; интерфейс цифрового ...

Sound processing device and sound processing method

A sound processing apparatus ( 100 ), which can improve precision of analyses on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors (first sound collector 110 - 1 and second sound collector 110 - 2 ), and the sound processing apparatus ( 100 ) is provided with a level signal conversion section (first level signal conversion section 130 - 1, second level signal conversion section 130 - 2 ) that converts the collected sound signal into a level signal, from which phase information is removed, a level signal synthesizing section ( 140 ) that generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors (first sound collector 110 - 1 and second sound collector 110 - 2 ) are synthesized, and a detecting and identifying section ( 160 ) that carries out analysis on the ambient sounds based upon the synthesized level signal.

Noise canceller and noise cancellation program

A directivity control unit 10 calculates a main beam signal with its directivity turned toward an object sound direction and a sub-beam signal with its blind spot turned toward the object sound direction from output signals of a plurality of microphones 2 and 3 through signal processing, and a frequency analyzing unit 20 converts them to spectra. A sound source decision unit 30 decides on whether a sound source is voice, stationary noise or unstationary noise from the spectra of the main beam signal and sub-beam signal and outputs as a sound source decision result, and calculates the average spectrum which is a statistic of noise for the main beam signal. An interfering sound removing unit 50 subtracts the average spectrum from the spectrum of the main beam signal to remove interfering sounds.

Dacs actuator

A compensation system for an implantable actuator is disclosed where the implantable actuator includes a sealed housing containing a driving arrangement for the actuator. The compensation system includes an external pressure sensor for measuring an external pressure outside of the sealed housing and a compensation module for determining a compensation factor for the implantable actuator based on the external pressure. In one embodiment, the compensation is directed to a direct acoustic cochlear stimulation (DACS) implantable actuator.

Method and apparatus of adjusting distribution of spatial sound energy

Provided is a method of adjusting a distribution of spatial sound energy, including storing information associated with a sound transfer function from each of speakers of a speaker array to a position of at least one listener, and information associated with the sound transfer function from each of the speakers of the speaker array to a far-field position, and generating at least two sound beams maximizing a far-field sound pressure attenuation with respect to a source signal, based on information associated with the sound transfer function, in order to form a personal sound zone in the position of the at least one listener.

Speaker device

A speaker device includes a signal processing section and a speaker array that includes a first speaker units group and a second speaker units group. The signal processing section performs a first signal process in which sound based on an audio signal of a first channel is set to output from the first speaker units group and a first virtual focus of the sound is set to an opposite side of the speaker array opposite to a sound output direction of the first speaker units group, and sound based on an audio signal of a second channel is set to output from the second speaker units group and a second virtual focus of the sound based on the audio signal of the second channel is set to an opposite side of the speaker array opposite to a sound output direction of the second speaker units group.

Ray tracing apparatus and method

The ray tracing apparatus for three-dimensional ( 3 D) graphics includes a Central Processing Unit (CPU) for constructing a first Acceleration Structure (AS) for a static object, and creating a second dynamic object by performing a Level Of Detail (LOD) operation on the first dynamic object and a ray tracing core for performing ray tracing based on the first AS and a second AS for the second dynamic object. The CPU or the ray tracing core constructs the second AS for the second dynamic object.

Multimedia Telecommunication Apparatus With Motion Tracking

A docking system for a personal communication terminal includes a base and a motorized mount joining the dock to the base and configured to rotate the dock about a vertical axis in response to a pan signal and about a horizontal axis in response to a tilt signal. The docking system further comprises a sensor array to produce signals indicative of the location of a user, a processor to convert the sensor output signals to tracking signals, and a controller to convert the tracking signals to pan and tilt signals, thereby to aim a camera.

Methods and apparatus for wireless microphone synchronization

Methods and apparatus related to timing synchronization used in a wireless microphone system to reduce and/or minimize interference are described. A wireless microphone receiver in some embodiments performs open loop timing control based on signals from an adjacent wireless microphone receiver and participates in closed loop timing control of wireless microphone transmitters which communicate with the particular wireless microphone receiver. The closed loop timing control includes instructing individual wireless microphone transmitters to advance or retard their symbol timing so that symbols transmitted by different wireless microphone transmitters are received in a synchronized manner at the wireless microphone receiver. The closed loop timing control allows a wireless microphone receiver to control wireless microphone transmitters to take into consideration their different distances to the receiver and the differing transmission times associated with the wireless microphone transmitters at different locations.

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.

Handheld electronic device with microphone array

A handheld electronic device includes a body and a microphone array. The body includes a side and a recess formed on the side. The microphone array includes a first microphone and a second microphone. Either the first and second microphones are disposed in the recess, or the first microphone is disposed in the recess while the second microphone is disposed outside the recess.

Slotted waveguide for loudspeakers

A slotted waveguide for use with an electrodynamic transducer is described. The slotted waveguide has a transducer front surface for radiating acoustic energy from the electrodynamic transducer. The transducer front surface is concave and the electrodynamic transducer is operable to a desired high frequency. The slotted waveguide may include a body and a volume displacement element. The slotted opening has a width not substantially greater than a wavelength corresponding to the desired high frequency and the volume displacement element has a displacement surface that extends inward into the electrodynamic transducer and complements in shape the transducer front surface.

Voice Volume Modulator

A small, compact voice volume monitor has been developed that provides feedback to the speaker/user regarding the volume of the user's speech. The monitor is based on a sensing sound vibrations in the ear bone during speech and converting these vibrations into an electrical signal reflecting the speech volume. Electronic circuitry is then used to compare the intensity of this signal with pre-set reference levels. When the intensity of the signal is outside the reference levels for a set amount of time, feedback is provided to the user, for example, from a small vibratory motor.

Sound processing apparatus and recording medium storing a sound processing program

A sound processing apparatus includes a first calculator that calculates first power based on a first signal received by a first microphone that is among the first microphone and a second microphone; a second calculator that calculates second power based on a second signal received by the second microphone; a gain calculator that calculates a gain on the basis of the ratio of the first power to the second power; and a multiplier that processes the second signal using the gain calculated by the gain calculator.

Active noise-cancellation feedback signal transmission

A noise-cancelling system includes headset for generating a feedback signal for noise-cancellation in response to sound externally generated from the headset. An encoded microphone signal is generated in response to the first feedback signal. An audio generator can be used to generate a noise-cancellation signal in response to the encoded microphone signal and to generate an electronic audio signal in response to the encoded microphone signal and a first output audio signal. An audio connector is provided to couple the encoded microphone signal from the headset to the audio generator and to couple the first electronic audio signal to the headset.

Control of a loudspeaker output

A loudspeaker drive circuit comprises an input for receiving an audio signal and a signal processor for processing the audio signal before application to the loudspeaker. The signal processor processes the audio signal to derive a loudspeaker drive signal which results in the loudspeaker membrane reaching its maximum displacement in both directions of diaphragm displacement.

Multiple Microphone System

A microphone system has a primary microphone for producing a primary signal, a secondary microphone for producing a secondary signal, and a selector operatively coupled with both the primary microphone and the secondary microphone. The system also has an output for delivering an output audible signal principally produced by one of the to microphones. The selector selectively permits either 1) at least a portion of the primary signal and/or 2) at least a portion of the secondary signal to be forwarded to the output as a function of the noise in the primary signal.

Audio coding device, audio coding method, and computer-readable recording medium storing audio coding computer program

An audio coding device includes a time frequency transform unit that, with respect to each of a plurality of channels included in an audio signal, generates a time frequency signal indicating frequency components at each time by performing a time frequency transform on a signal of the channel; a transient detection unit that detects a transient with respect to each of the plurality of channels so as to obtain a transient detection time; a transient time correction unit that, when a difference in transient detection times between an early detection channel in which the transient detection time is earliest and a late detection channel that is a channel other than the early detection channel among the plurality of channels is within a range in which the transient; a grid determination unit that, with respect to each of the plurality of channels, and a coding unit that codes.

Accelerometer vector controlled noise cancelling method

A telecommunication device is disclosed, comprising: a microphone array comprising a plurality of microphones, wherein each microphone receives an analogue acoustic signal; a position sensing device for determining how the telecommunication device is positioned in three-dimensions with respect to a user's mouth; at least one analogue/digital converter for converting each analogue acoustic signal into a digital signal; a digital signal processor for performing signal processing on the received digital signals comprising a controller, a plurality of delay circuits for delaying each received signal based on an input from the controller and a plurality of preamplifiers for adjusting the gain of each received signal based on a gain input from the controller, wherein the controller selects the appropriate delay and gain values applied to each received signal to remove noise from the received signals based on the determined position of the telecommunication device. A method for creating and controlling a location of a virtual microphone near a telecommunication device so as to reduce background noise in a speech signal is also disclosed.

Surface-Mounted Microphone Arrays on Flexible Printed Circuit Boards

A microphone array, having a three-dimensional (3D) shape, has a plurality of microphone devices mounted onto (at least one) flexible printed circuit board (PCB), which is bent to achieve the 3D dimensional shape. Output signals from the microphone devices can be combined (e.g., by weighted or unweighted summation or differencing) to form sub-element output signals and/or element output signals, and ultimately a single array output signal for the microphone array. The PCB may be uniformly flexible or may have rigid sections interconnected by flexible portions. Possible 3D shapes include (without limitation) cylinders, spirals, serpentines, and polyhedrons, each formed from a single flexible PCB. Alternatively, the microphone array may be an assembly of multiple, interconnecting sub-arrays, each having two or more rigid portions separated by one or more flexible portions, where each sub-array has at least one cut-out portion for receiving a rigid portion of another sub-array.

Microphone

A microphone includes a plurality of microphone units; in which the microphone units include a first group of microphone units and a second group of microphone units, the first group of microphone units and the second group of microphone units are disposed alternately, the first group of microphone units are connected in series such that outputs from the first group of microphone units are added and outputted as an added output, the second group of microphone units are connected in series such that outputs from the second group of microphone units are added and outputted as another added output, and the added output of one of the first group of microphone units and the second group of microphone units is output from a hot terminal as a balanced output and the other added output is output from a cold terminal as a balanced output.

Adaptive noise canceling architecture for a personal audio device

A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal that measures the ambient audio and an error microphone signal that measures the output of an output transducer plus any ambient audio at that location and injects the anti-noise signal at the transducer output to cause cancellation of ambient audio sounds. A processing circuit uses the reference and error microphone to generate the anti-noise signal, which can be generated by an adaptive filter operating at a multiple of the ANC coefficient update rate. Downlink audio can be combined with the high data rate anti-noise signal by interpolation. High-pass filters in the control paths reduce DC offset in the ANC circuits, and ANC coefficient adaptation can be halted when downlink audio is not detected.

Active Delay Method and a improved wireless binaural hearing device using the same method

Disclosed are an active delay method and an improved wireless binaural hearing device. The binaural hearing device includes: a first hearing device including a first microphone, an amplifier and a wireless transmitter; and a second hearing device including a second microphone, an amplifier, a wireless transmitter, a wireless receiver which receives a signal from the wireless transmitter, an active delay circuit which synchronizes the received signal with a signal acquired by the second microphone, a neural network which synchronizes the delayed signal, and a speaker which converts the synchronized signal into a voice signal. With this configuration, it is possible to prevent incorrect detection of the position of the sound source or paralalia due to a time delay which is produced in the wireless binaural hearing device and reduce noises due to a time difference between both hearing devices, thereby providing a binaural hearing device with high quality.

Horn enclosure for combining sound output

A horn enclosure combines sound output emanating from a plurality of acoustic drivers. The horn enclosure includes a plurality of sound input plenums, a plurality of flow passageways, and a common sound output chamber. Each of the input plenums is coupled to an acoustic driver for receiving generated sound output from the associated driver. A plurality of partitions defining the plurality of flow passageways provides for a plurality of acoustic paths from each acoustic driver and through the input plenum. The common sound output chamber is in acoustic communication with each plenum through respective flow passageways and has a sound output opening for passage of the combined sound output.

Switching amplifier and audio device

A switching amplifier is provided. The switching amplifier includes an input unit which receives an audio signal; a first switching device and a second switching device which switch and output the audio signal; a first snubber circuit which is commonly connected to the first and second switching devices and which reduces overshoot of the audio signal that is output by the first and second switching devices via switching; and a second snubber circuit which is commonly connected to the first and second switching devices, which is connected in parallel to the first snubber circuit and which reduces the overshoot of the audio signal, wherein the first and second snubber circuits alternately reduce the overshoot of the audio signal.

System, Method, and Computer Program Product for Voice Decibel Monitoring on Electronic Computing Devices

The present invention in general relates to a voice decibel application and a method for a phone and in particularly relates to the application and the method for a phone to measure the voice decibel during a phone call. The application incorporated into the user's phone is configured to measure the voice decibel during a phone call and send a signal to the user in case the voice decibel exceeds or goes below a preset voice decibel threshold level. The present invention is advantageous because it will help the users to recognize their talking habits and assist them in improving their talking habits.

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.

Array microphone apparatus for generating a beam forming signal and beam forming method thereof

Embodiments described in the present disclosure relate to an array microphone apparatus for generating a beam forming signal. The apparatus includes first, second, and third omni-directional microphones, each converting an audible signal into a corresponding electrical signal. The second omni-directional microphone is disposed between the other two omni-directional microphones. The apparatus includes a first directional microphone forming device to jointly output a first directional microphone signal with a first bi-directional pattern, and a magnitude and phase response handler device to output a second directional microphone signal with an omni-directional pattern shifted by a prefixed value with respect to first directional microphone signal. The apparatus further includes a combining device receiving the first and second directional microphone signals and outputting a combined directional microphone signal with a combined beam pattern correlated to the first bi-directional and second omni-directional patterns, the combined directional microphone signal being in a broadside configuration.

Oscillator and electronic device

In an oscillator ( 100 ), a plurality of piezoelectric vibrators ( 111 to 113 ) supported by vibrator support mechanisms ( 120 ) individually output highly directional sound waves. The plurality of piezoelectric vibrators ( 111 to 113 ) is formed by dividing a laminated body of an elastic member and a piezoelectric substance by the vibrator support mechanisms ( 120 ). Since it is not necessary to arrange the plurality of piezoelectric vibrators in a matrix, the entirety of a device can be small-sized. A sound deflection unit that deflects a sound wave which is output by at least one of the piezoelectric vibrators may be further included.

Digital signal generator and digital microphone

According to one embodiment, a digital signal generator includes an amplifying unit, a reference voltage generator and a modulator. The amplifying unit amplifies an analog input signal having a signal level linearly depending on a temperature. The reference voltage generator generates a reference voltage linearly depending on the temperature. The modulator converts the analog input signal amplified by the amplifying unit into a digital output signal based on the reference voltage.

Information processing apparatus and information processing method

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

Decisions on ambient noise suppression in a mobile communications handset device

A method in a communications device includes the following operations. During a call, a process automatically detects that the device has moved from an at-the-ear position to an away-from-the-ear position. Based on the detection, a noise suppressor that operates upon an uplink signal for the call is signaled to change its noise suppression performance. Other embodiments are also described and claimed.

Speech enhancement method

A speech enhancement method is disclosed. The method includes the steps of: receiving a plurality of frames of sound signals by a microphone array; calculating an inter-aural time difference for each frequency band of each frame of the sound signals corresponding to at least one two-microphone set of the microphone array; calculating a plurality of values of cumulative histograms according to the calculated inter-aural time differences; determining a first inter-aural time difference threshold according to the calculated value of the cumulative histograms; and filtering the plurality of frames of sound signals according to the first inter-aural time difference threshold.

Apparatus and method for suppressing noise from voice signal by adaptively updating wiener filter coefficient by means of coherence

A voice signal processor detects background noise sections to reflect characteristics of the background noise on the Wiener filter coefficient to be used for suppressing noise components of input voice signals. In the voice signal processor, directivity signal generators form directivity signals having a directivity pattern. The directivity signals are used by a coherence calculator to obtain coherence, which is in turn used by a targeted voice section detector to detect a targeted voice section. A background noise section detector detects background noise sections containing no voice signal. When a background noise section is detected, a WF adapter uses characteristics of background noise in the detected temporal section to calculate a new WF coefficient.

ACOUSTIC TRANSDUCER, AND MICROPHONE USING THE ACOUSTIC TRANSDUCER

An acoustic sensor includes: a semiconductor substrate; a vibrating membrane, formed above the semiconductor substrate, which includes a vibrating electrode; and a fixed membrane, formed on an upper surface of the semiconductor substrate, which includes a fixed electrode, the acoustic sensor detecting an acoustic wave according to a change in capacitance between the vibrating electrode and the fixed electrode. The fixed membrane has a plurality of sound hole portions formed therein in order to allow the acoustic wave to reach the vibrating membrane from outside, and the fixed electrode is formed so that a boundary of an edge portion of the fixed electrode does not intersect the sound hole portions. 1. An acoustic transducer comprising:a substrate;a vibrating membrane, formed above the substrate, which includes a vibrating electrode; anda fixed membrane, formed on an upper surface of the substrate, which includes a fixed electrode, said acoustic transducer converting an acoustic wave into an electrical signal according to a change in capacitance between the vibrating electrode and the fixed electrode,the fixed membrane having a plurality of sound hole portions formed therein in order to allow the acoustic wave to reach the vibrating membrane from outside,the fixed electrode being formed so that a boundary of an edge portion of the fixed electrode does not intersect the sound hole portions.2. The acoustic transducer according to claim 1 , whereinthe sound hole portions are regularly arrayed, andthe fixed electrode is in a shape having sides each extending along any one of the following directions: array directions of the sound hole portions; and directions each bisecting an angle formed by two array directions adjacent to each other among the array directions.3. The acoustic transducer according to claim 2 , wherein the fixed electrode is in a stepped shape in order to be similar in shape to a vibrating portion of the vibrating electrode.4. The acoustic transducer ...

LOUDSPEAKER WITH INTERLOCKING MAGNET STRUCTURE

A loudspeaker having a magnet system and a method of assembling the magnet system are disclosed. The magnet system comprises a magnet and an armature core that is mounted on the magnet. The magnet system also includes a shell pot configured to receive the magnet and the armature core in a hollow interior. The magnet system further includes a shaft that interlocks with the magnet, the armature core and the shell pot, and that is, on one end, mechanically connected to the shell pot. The magnet system also includes a push-on fastener that has an aperture through which the shaft passes and that is secured to the shaft at another end of the shaft such that it applies pressure onto the first surface of the armature core to fixedly position the armature core and the magnet with respect to the shell pot. 1. A loudspeaker having a magnet system , the magnet system comprising:a magnet that has a first surface, a second surface and an aperture;an armature core that has a first surface, a second surface and an aperture and that is mounted on the magnet, where the second surface of the armature core contacts the first surface of the magnet;a shell pot configured to receive the magnet and the armature core in a hollow interior, where the second surface of the magnet contacts a base surface of the shell pot;a shaft that interlocks with the magnet, the armature core and the shell pot, that extends through the aligned apertures included in each of the magnet, the armature core and the shell pot, and that is, on one end, mechanically connected to the shell pot; anda push-on fastener that has an aperture through which the shaft passes and that is secured to the shaft at another end of the shaft such that the push-on fastener applies pressure onto the first surface of the armature core to fixedly position the armature core and the magnet with respect to the shell pot.2. The loudspeaker of claim 1 , further comprising a further magnet that has a first surface claim 1 , a second surface ...

APPARATUS

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 change of position of the apparatus, and processing at least one audio signal dependent on the change in position. 1. A method comprising:determine a change of position of the apparatus;processing at least one audio signal dependent on the change in position.2. The method as claimed in claim 1 , wherein the change in position is at least one of:a relative change of position with respect to a further object; andan absolute change of position.3. The method as claimed in claim 1 , wherein the change in position comprises at least one of:a change in translational position; anda change in rotational position.4. The method as claimed in claim 1 , further comprising:detecting a first position of the apparatus;receiving at least one audio signal; andgenerating for each audio signal at least one signal processing parameter dependent on the first position of the apparatus.5. The method as claimed in claim 4 , wherein generating for each audio signal at least one signal processing parameter dependent on the first position of the apparatus comprises generating at least one of:gain; anddelay.6. The method as claimed in claim 4 , further comprising:generating for each audio signal at least one further signal processing parameter dependent on the detected change of position of the apparatus.7. The method as claimed in claim 6 , wherein the generating for each audio signal at least one further signal processing parameter comprises:determining whether the change of position of an apparatus is greater than at least one predefined value; andgenerating the at least one further signal processing parameter for each audio signal dependent on the at least one predefined value.8. The method as claimed in claim 1 , wherein processing the ...

Heat Dissipating Acoustic Transducer with Mounting Means

The inertial transducer comprises a foot, a modular key, at least one suspension means having at least three contact points, and heat dissipation means comprising multiple heat transfer points. The foot alternatively includes an opening or a cap. The cap may be frangible. Preferably, the foot includes a cylindrical wall that extends within the inertial transducer and the foot's lower surface is coplaner with the lower surface of a housing of the transducer, thereby reducing stack height. The cylindrical wall may or may not be threaded and may accept a shaft from a receiving apparatus used to associate the transducer with a substrate. 1. An inertial type acoustic transducer comprising a magnetic motor circuit comprising at least one of the group consisting of at least one yoke and at least one magnet; said components of the magnetic motor stacked and leaving a small magnetic gap bounded by an inner wall of the yoke , said motor further comprising a voice coil positioned at least partially in said magnetic gap.2. The inertial type acoustic transducer of wherein the magnetic motor further comprises a top plate.3. The inertial type acoustic transducer of further comprising a housing having an inside surface and a primary suspension claim 1 , wherein said primary suspension comprises associating said inside surface with the magnet motor.4. The inertial type acoustic transducer of wherein said inside surface of said housing comprises compliant material allowing for axial displacement of said magnetic motor circuit.5. The inertial type acoustic transducer of wherein said compliant material is also resilient.6. The inertial transducer of further comprising a foot structure generally covering said housing claim 3 , said foot structure comprising an opening claim 3 , an inside surface claim 3 , and an outer surface.7. The inertial transducer of wherein said opening comprises a diameter generally concentric with and substantially equal to a diameter of a voice coil former.8. ...

Satellite Microphones for Improved Speaker Detection and Zoom

Architecture for exploiting satellite microphones and employing other techniques of conference room camera/microphone systems to significantly improve the true positive rate (reduce false positives) in sound source localization (SSL). Techniques for realizing the improvement include using an LED emitter to determine the precise location of the satellite microphones on a table, using the base SSL and external sounds to determine the approximate location of the satellite microphone on the table, using the satellite microphone phase to improve the SSL performance, using the satellite microphone amplitude to improve the active speaker detector (ASD) performance, and using the satellite microphones to estimate camera zoom. 1. A method of detecting a speaker comprising:extending audio range of a local microphone array of a conferencing base station using satellite microphones;computing location information for location of the satellite microphones relative to the conferencing base station; andprocessing the location information and audio signals of the local microphone array and the satellite microphones for automatic detection of a speaker relative to the conferencing base station.2. The method of claim 1 , further comprising locating the satellite microphones on opposing regions of the conferencing base station according to corresponding nominal locations and estimating location of a satellite microphone relative to the regions based on audio power.3. The method of claim 1 , further comprising uniquely identifying each of the satellite microphones using emitters that transmit unique wireless signals for capture by a camera system of the conferencing base station and image processing to identify the satellite microphones.4. The method of claim 1 , further comprising estimating the location distances of the satellite microphones from the conferencing base station based on camera images of the satellite microphones claim 1 , vertical field of view of a camera claim 1 , and ...

SOUND GENERATION SYSTEM, ULTRASONIC WAVE EMITTING DEVICE, AND ULTRASONIC WAVE EMITTING METHOD

Disclosed are a sound generation system, an ultrasonic wave emitting device, and an ultrasonic wave emitting method which utilize the principle of parametric speakers (which uses the difference between frequencies of two ultrasonic waves) and which are capable of allowing target persons in a prescribed area to hear a sound only when a mobile body approaches said prescribed area. An alert system () has an ultrasonic wave emitting device () mounted on a mobile body (), and an ultrasonic wave emitting device () permanently installed near an area (A). The ultrasonic wave emitting device () emits a first ultrasonic wave towards an area at least including the aforementioned area (A), and meanwhile, the ultrasonic wave emitting device () emits a second ultrasonic wave. The ultrasonic emitting device () forms the second ultrasonic wave by means of modulating the ultrasonic wave of frequency fit with an audible signal. 110-. (canceled)12. The ultrasound emitting apparatus according to claim 11 , further comprising an audible signal controlling section that varies contents of the audible signals in accordance with the speed of the moving object.13. The ultrasound emitting apparatus according claim 11 , further comprising:a reception section that receives information on a reference frequency of ultrasound signal that are transmitted from another ultrasound emitting apparatus and are used by the other ultrasound emitting apparatus; anda frequency controlling section that adjusts a frequency of the ultrasound signals to be the same frequency as the reference frequency of the other ultrasound emitting apparatus.14. The ultrasound emitting apparatus according to claim 11 , further comprising:a reception section that receives frequency information on a frequency associated with a traffic lane in an intersection, from another ultrasound emitting apparatus placed close to the intersection; anda frequency controlling section that adjusts a frequency of the ultrasound signals to be a ...

POWER MANAGEMENT CONTROLLER FOR DRIVERS

A driver controller comprising a state machine for controlling transitions between a plurality of states. An output switch for providing a low impedance path to ground during transition periods. An output stage for decoupling output signal from driver amplifier during the transition periods. 114-. (canceled)15. A method for controlling a driver comprising:determining a first audio driver state based on a plurality of first driver state variables;receiving one or more second state variables; andtransitioning from the first audio driver state to one of a plurality of second audio driver states in response to the one or more second state variables.16. The method of wherein determining the first audio driver state based on the plurality of first driver state variables comprises determining an audio driver reset state based on an amplifier state variable and an output stage state variable.17. The method of wherein determining the first audio driver state based on the plurality of first driver state variables comprises determining an audio driver mute state based on an output switch state variable and an output stage state variable.18. The method of wherein determining the first audio driver state based on the plurality of first driver state variables comprises determining an enabled audio driver state based on a driver state variable and a digital to analog converter state variable.19. The method of further comprising:receiving one or more third state variables; andtransitioning from the one of the plurality of second audio driver states to another of the plurality of second audio driver states in response to the one or more third state variables.20. The method of further comprising:receiving one or more third state variables; andtransitioning from the one of the plurality of second audio driver states to one of the first driver state or another of the plurality of second audio driver states in response to the one or more third state variables.21. A method for ...

Method and device for phase-sensitive processing of sound signals

A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector φ0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F.

AUDIO MIXING PROCESSING METHOD AND APPARATUS FOR AUDIO SIGNALS

An embodiment of the present disclosure provides an audio mixing processing method for audio signals. The method includes: determining a terminal that needs to adjust a location of an audio signal, and adjusting location information about the audio signal for the terminal; and performing audio mixing processing for the audio signal whose location is adjusted and other signals to be audio mixed. An embodiment of the present disclosure further provides an audio mixing processing apparatus for audio signals. According to the solutions provided in the embodiments, the audio location information about terminals whose audio signal locations overlap are adjusted, so that the location of each sending terminal is separated from each other as far as possible, and the sound of each sending terminal is more clear, thereby improving the field experience and feeling of a user. 1. An audio mixing processing method , comprising:determining a terminal that needs to adjust a location of an audio signal, and adjusting location information about the audio signal for the terminal; andperforming audio mixing processing for the audio signal whose location is adjusted and other signals to be audio mixed.2. The method according to claim 1 , wherein determining the terminal that needs to adjust the location of the audio signal comprises determining the terminal when one of the following conditions is satisfied:when the locations of the audio signals of the terminal and other terminals overlap;when the location of the audio signal of the terminal does not match the position of the terminal in a video picture of multiple pictures; andwhen the terminal participates in audio mixing for the first time3. The method according to claim 1 , wherein when multiple terminals whose locations are to be adjusted exist claim 1 , determining a terminal that needs to adjust a location of an audio signal comprises: adjusting the location information about audio signals of the multiple terminals whose locations ...

MOBILE WIRELESS COMMUNICATIONS DEVICE WITH REDUCED MICROPHONE NOISE FROM RADIO FREQUENCY COMMUNICATIONS CIRCUITRY

A mobile wireless communications device includes a housing and circuit board in the housing and having radio frequency (RF) circuitry and a power amplifier and microphone mounted thereon. An antenna is carried within the housing and operative with the RF circuitry. An RF shield surrounds and isolates the microphone from the RF circuitry, power amplifier and antenna and shields the microphone from radiated energy generated from the RF circuitry, antenna or power amplifier. 124-. (canceled)25. An electronic device comprising:radio frequency (RF) circuitry;a microphone susceptible to RF interference from the RF circuitry;an RF shield surrounding the microphone;a cover adjacent the RF shield;an acoustical tube extending between the RF shield and cover; andan electrically conductive ring surrounding the acoustical tube and extending between opposing surfaces of the RF shield and cover.26. The electronic device according to claim 25 , further comprising a circuit board mounting the RF circuitry claim 25 , microphone and RF shield.27. The electronic device according to claim 25 , further comprising a power amplifier coupled to the RF circuitry.28. The electronic device according to claim 25 , further comprising an antenna coupled to the RF circuitry.29. The electronic device according to claim 28 , wherein the antenna comprises a dielectric substrate and at least one conductive trace thereon.30. The electronic device according to claim 25 , wherein the RF shield and the cover each comprises metal.31. The electronic device according to claim 25 , wherein the RF circuitry comprises a transceiver chip set.32. The electronic device according to claim 25 , wherein the RF circuitry is operative for generating Global Systems for Mobile (GSM) packet bursts.33. The electronic device according to claim 25 , further comprising an isolation compartment containing the RF circuitry.34. The electronic device according to claim 25 , wherein the microphone comprises a surface mounted ...

Multi-functional weather alert communication key chain

The present invention encompasses a key chain device which receives disaster and/or radio broadcast worldwide. The digital component scans for and receives, the strongest broadcast signal, within the vicinity of the user and/or where disaster is occurring. Comprised of both digital & analog receivers and transmitters, the device can furthermore be used to communicate with and control analog and digital devices. This programmable feature enables the device to perform such functions as remotely controlling car starting/alarms, garage doors, and house alarms, etc. Enabling the internal microprocessor, said device can serve to entertain users with the digital media player, keeping them connected with a GPS locater. Additionally, the present invention has a cellular telephone component, with the ability to pair with wireless Bluetooth devices. This invention is a novel use of components, providing users with an enhanced level of control, convenience, electronic device connectivity and interaction in their daily life. 1. A Multi-functional Weather Alert Communication Key Chain assembly , comprising: a deployable ergonomic housing adapted to be transported by end user , said housing comprising one half , with indentations to accommodate a screen display element , a second bottom half , with indentations to accommodate a solar panel auxiliary power source with spacing between both halves accommodating tactile functional control elements or buttons , which further control the power on/off means , with an external portion of the device thereof adapted to be coupled to a removable keychain eyelet , said device furthermore comprising a central microprocessor unit , an analog signal processor means , a digital signal processor means , a hardline charging/data transfer connection port , an internal rechargeable battery or power supply means , a Bluetooth™ or wireless data transfer connection module , a motorized vibrational element within said housing , an internal analog or ...

SETTING TELEVISION OR RADIO DEFAULT CHANNEL

In some embodiments, a media device is configured so that it automatically causes a first predetermined channel to be displayed on an output device when a user sends a request signal to the media device to turn the output device off. The media device may be configured so that it causes the output device to shut off after the first predetermined channel has been on for a dwell period of time after the request signal has been received. The dwell period of time may be approximately two seconds. In some embodiments, a media device is configured so that it automatically causes a first predetermined channel to be displayed on an output device when a user sends a request signal to the media device to turn the output device on. In some embodiments, the media device is a television device. In other embodiments, the media device is a radio device. 1. A method comprising:a radio device receiving a first request to turn the radio device off;the radio device automatically switching to a first predetermined radio channel in response to receiving the first request; andthe radio device automatically causing at least one radio speaker of the radio device to output the first predetermined radio channel when the media device receives the first request to turn the radio device off such that one or more sounds currently being emitted on the first predetermined radio channel are emitted on the radio speaker, in response to the radio device receiving the first request, for a predetermined dwell period of time after the first request is received by the radio device prior to the radio speaker turning off.2. The method of further comprising the radio device receiving a first signal from a default channel controller to set the first predetermined channel.3. The method of further comprising the radio device receiving a first signal from a default channel controller to set the predetermined dwell period of time.4. A method comprising:a radio device receiving a first request to turn the radio ...

Microphone Device, Microphone System and Method for Controlling a Microphone Device

There is provided a microphone device comprising a camera having a field of vision for acquiring image data, a microphone unit with adjustable directivity and a control unit for adjusting the directivity of the microphone unit. Adjustment of the directivity of the microphone unit is based on ascertained position information of at least one user in the field of vision of the camera. The camera and/or the control unit are adapted to ascertain position information of at least one user from the image data acquired by the camera. 1. A microphone device comprising;at least one camera having at least one field of vision for acquiring image data;at least one microphone unit having at least one adjustable directivity; anda control unit configured to adjust the directivity of the at least one microphone unit based on ascertained position information of at least one user in the field of vision of the at least one camera;wherein at least one of the camera and the control unit is adapted to ascertain position information of at least one user from the acquired image data; andwherein the control unit is adapted to control focusing of the directivity of the microphone unit in accordance with the size of an acquired image portion based on face recognition.2. The microphone device as set forth in ;wherein the position information of the user is ascertained based on face recognition from the acquired image data of the camera.3. The microphone device as set forth in ;wherein the control unit is adapted to control the directivity of the microphone unit so that there is more than one main direction of directivity if more than one user has been detected on the basis of the ascertained position information.4. The microphone device as set forth in ;wherein the control unit is adapted to mute the output audio signal in dependence on the acquired audio and/or video signals.5. A microphone system comprising: at least one camera having at least one field of vision for acquiring image data;', ' ...

Headset Systems And Related Methods

In one embodiment, a headset system comprises a handsfree headset comprising a headset docking portion and an earpiece, a docking station comprising a docking mechanism configured to hold the handsfree headset by the headset docking portion, and a magnetic securing mechanism configured to magnetically secure the handsfree headset to the docking station. The handsfree headset can be configured to receive signals from an electronic device, be carried in a handsfree manner aligning the earpiece and an ear of a user when in use, and be held by the docking station independently of the electronic device when not in use. Other examples and embodiments are described herein. 1. A headset system comprising:a handsfree headset comprising a headset docking portion and an earpiece;a docking station comprising a docking mechanism configured to receive the headset docking portion of the handsfree headset; anda magnetic securing mechanism configured to magnetically secure the handsfree headset to the docking station; receive signals from an electronic device;', 'be carried in a handsfree manner aligning the earpiece and an ear of a user when in use; and', 'be received by the docking station independently of the electronic device when not in use., 'wherein the handsfree headset is configured to2. The headset system of claim 1 , wherein:the docking mechanism is configured to bound the headset docking portion when the docking station holds the handsfree headset.3. The headset system of claim 1 , wherein:the earpiece is insertable inside a concha of the ear to restrict movement of the handsfree headset relative to the ear.4. The headset system of claim 1 , wherein: a first magnetic element at the docking mechanism of the docking station; and', 'a second magnetic element at the headset docking portion of the handsfree headset., 'the magnetic securing mechanism comprises5. The headset system of claim 1 , wherein:the earpiece is located outside of the docking station when the headset ...

MICROPHONE

A microphone comprises a sound inlet, a first MEMS microphone structure comprising a first membrane and associated backplate in the path of the sound inlet and a second MEMS microphone structure comprising a second membrane and associated backplate stacked with respect to the first microphone membrane, with an enclosed space between the first and second membranes. 1. A microphone comprising:a sound inlet;a first microphone structure comprising a first membrane and associated backplate in the path of the sound inlet;a second microphone structure comprising a second membrane and associated backplate stacked with respect to the first microphone structure, with an enclosed space between the first and second membranes,wherein the microphone structures are back to back, such that movement of one membrane towards its associated backplate is accompanied by movement of the other membrane away from its associated backplate; anda signal processor for combining microphone signals of the first and second microphone structures to generate the microphone output; a non-linear microphone signal response of one of the microphone structures; and', 'pull-in of the membranes of one or both of the microphone structures., 'characterized in that the signal processor is adapted to combine the microphone signals of the first and second microphone structures to detect at least one of2. (canceled)3. (canceled)4. (canceled)5. (canceled)6. A microphone as claimed in claim 1 , wherein the signal processor is adapted to determine the presence of a non-linear microphone signal response based on a threshold applied to the combined microphone signals.7. A microphone as claimed in claim 1 , further comprising means for:adjusting the microphone electrical load; orselecting different parts of the microphone signals of the first and second microphone structures to derive the microphone output signal,based on the detection of the non-linear microphone signal response.8. (canceled)9. (canceled)10. A method ...

Method and device of channel equalization and beam controlling for a digital speaker array system

A method and device of channel equalization and beam controlling for a digital speaker array system includes (1) converting digital format; (2) performing channel equalization; (3) controlling beam-forming; (4) performing multi-bit Σ-Δ modulation; (5) performing thermometer code conversion; (6) performing dynamic mismatch-shaping processing; and (7) extracting the channel information to send to the digital power amplifier and drive the array sound. A device includes a sound source, a digital converter, a channel equalizer, a beam-former, a Σ-Δ modulator, a thermometer coder, a dynamic mismatch shaper, an extraction selector, a multi-channel digital power amplifier and a speaker array. Each unit connects to each other serially.

PIEZOELECTRIC MICROPHONES

Electronic devices and microphone devices are described. 1. An electronic device , comprising:a first microphone operative to receive audio signals from a first, direction;a second microphone operative to receive audio signals from a second direction; anda controller operative to engage selectively the second microphone to receive ambient audio noise or to receive an audio input.2. An electronic device as claimed in claim 1 , further comprising a microprocessor adapted to receive an output signal from the second microphone and operative to provide noise cancellation to an output signal of the first microphone.3. An electronic device as claimed in claim 1 , wherein the audio input is an audio portion of an audio/video signal.4. An electronic device as claimed in claim 1 , wherein the first microphone is adapted to receive a voice input.5. An electronic device as claimed in claim 1 , further comprising at least one additional microphone claim 1 , which is adapted to receive ambient audio noise or to receive an audio input claim 1 , or both.6. An electronic device as claimed in claim 1 , wherein at least one of the microphones is a piezoelectric microphone.7. An electronic device as claimed in claim 1 , wherein the first microphone comprises a first film bulk acoustic (FBA) structure and the second microphone comprises a second FBA structure.8. An electronic device claim 1 , comprising:a first microphone operative to receive audio signals from a first, direction;a second microphone operative to receive audio signals from a second direction; anda controller operative to engage selectively the second microphone to receive ambient audio noise or to receive an audio input, wherein the controller is adapted to receive an output signal from the second microphone and operative to provide noise cancellation to an output signal of the first microphone.9. An electronic device claim 1 , comprising:a first microphone operative to receive audio signals from a first direction;a ...

MICROPHONE ARRAY CONFIGURATION AND METHOD FOR OPERATING THE SAME

An apparatus comprises a plurality of microphone units including at least a first microphone unit and a second microphone unit, each of the first and second microphone units comprising a microphone, an analog-to-digital converter, and a local memory. The microphone is configured to capture an analog audio signal. The analog-to-digital converter is configured to convert the analog audio signal created by the microphone into a digital audio signal. The local memory is configured to store the digital audio signal. The apparatus further comprises, a controller unit comprising a processor configured to process the digital audio signals. The first microphone unit and the second microphone unit are operatively connected to the controller unit in a series configuration, the second microphone unit being configured to output the digital audio signal to the first microphone unit, and the first microphone unit being configured to output the digital audio signal to the controller unit. 1. An apparatus comprising:a plurality of microphone units including at least a first microphone unit and a second microphone unit, each of the first and second microphone units comprising a microphone, an analog-to-digital converter, and a local memory, wherein the microphone is configured to capture an analog audio signal, the analog-to-digital converter is configured to convert the analog audio signal created by the microphone into a digital audio signal, and the local memory is configured to store the digital audio signal; anda controller unit comprising a processor configured to process the digital audio signals,wherein the first microphone unit and the second microphone unit are operatively connected to the controller unit in a series configuration, the second microphone unit being configured to output the digital audio signal to the first microphone unit, and the first microphone unit being configured to output the digital audio signal to the controller unit.2. The apparatus of claim 1 , ...

MULTICHANNEL ACOUSTIC ECHO REDUCTION

A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal. 1. A method executed by a processor of a computing device , the method comprising:transmitting a first calibration signal to a first speaker, the first calibration signal causing the first speaker to generate first output over a first time period;transmitting a second calibration signal to a second speaker, the second signal causing the second speaker to generate second output over a second time period that is non-overlapping with the first time period;receiving a first microphone signal from a first microphone, the first microphone signal corresponding to the first output from the first speaker over the first time period;receiving a second microphone signal from the first microphone, the second microphone signal corresponding to the second output from the second speaker over the second time period;computing a first coefficient of a first fixed filter for the first microphone based upon the first calibration signal and the first microphone signal;computing a second coefficient of a second fixed filter for the first microphone based upon the second calibration signal and the second microphone signal; andsubsequent to computing the first coefficient of the first fixed filter and the second coefficient of the second fixed filter, applying the first fixed filter and the second fixed filter to an acoustic signal captured by the first microphone.2. The method of claim 1 , wherein applying ...

VOICE ACQUISITION SYSTEM FOR VEHICLE

A voice acquisition system for a vehicle includes a microphone having a first microphone element and a second microphone element. The system includes circuitry for processing signals output by the first and second microphone elements. The first microphone element is disposed at a first acoustic port and the second microphone element is disposed at a second acoustic port. The first acoustic port is at least partially isolated from the second acoustic port. When the microphone is normally mounted in an interior cabin of a vehicle, vocal signals emanating from an occupant of the vehicle pressurize the first microphone element at the first acoustic port ahead of pressurization by the vocal signals of the second microphone element at the second acoustic port. The voice acquisition system may include at least one of (i) discrimination of vocal signals from non-vocal signals, (ii) noise cancellation and (iii) filtering of noise. 1. A voice acquisition system for a vehicle , said voice acquisition system comprising:a microphone;wherein said microphone comprises a first microphone element and a second microphone element;circuitry for processing signals output by said first and second microphone elements;wherein said first microphone element is disposed at a first acoustic port and wherein said second microphone element is disposed at a second acoustic port;wherein said first acoustic port is at least partially isolated from said second acoustic port;wherein, when said microphone is normally mounted in an interior cabin of a vehicle equipped with said voice acquisition system, vocal signals emanating from an occupant of the equipped vehicle pressurize said first microphone element at said first acoustic port ahead of pressurization by said vocal signals of said second microphone element at said second acoustic port; andwherein said voice acquisition system comprises at least one of (i) discrimination of vocal signals from non-vocal signals, (ii) noise cancellation and (iii) ...

SOUND SOURCE SIGNAL FILTERING APPARATUS BASED ON CALCULATED DISTANCE BETWEEN MICROPHONE AND SOUND SOURCE

Provided is a sound source signal filtering method and apparatus. The sound source signal filtering method includes: generating two or more microphone output signals by combining sound source signals input through a plurality of microphones; calculating distances between the microphones and a sound source from which the sound source signals are emitted by using distance relationships according to frequencies of the sound source signals extracted from the generated microphone output signals; and filtering the sound source signals to obtain one or more sound source signals corresponding to a predetermined distance by using the calculated distances. Accordingly, it is possible to obtain only sound source signals emitted from a sound source at a particular distance from the microphone array among a plurality of sound source signals input through the microphone array. 1. A sound source signal filtering apparatus comprising:a microphone output signal generator generating two or more microphone output sound signals by combining sound source signals input through a plurality of microphones;a distance calculator calculating distances from the microphones to a sound source from which the sound source signals are emitted, by using distance relationships based on frequencies of the sound source signals of the generated microphone output sound signals; anda signal filter filtering the sound source signals to obtain one or more sound source signals corresponding to a predetermined distance by using the calculated distances.2. The apparatus of claim 1 , further comprising an operator generating result values independent from magnitudes of the generated microphone output sound signals by removing the magnitudes claim 1 ,wherein the distance calculator calculates a distance between the computation result value to the sound source.3. The apparatus of claim 1 , wherein the microphone output sound signals have different sound attenuation rates from each other based on the calculated ...

AUDIO AMPLIFICATION CIRCUIT

Disclosed is an audio amplification circuit comprising: an input terminal for receipt of an audio input signal; a first preamplifier having an input operatively coupled to the input terminal and operable to provide a first amplified audio signal with a first signal amplification; a second preamplifier having an input operatively coupled to the input terminal and operable to provide a second amplified audio signal with a second signal amplification, smaller than the first signal amplification; a switch having a first input operatively coupled to the first preamplifier, a second input operatively coupled to the second preamplifier, and an output; an analogue-to-digital converter operatively coupled to the output of the switch and operable to provide a digital audio signal; a signal selection circuit operable to control the switch to selectively provide one of the first and second amplified audio signals on the output of the switch. 1. An audio amplification circuit comprising:a first preamplifier having an input operatively coupled to an input terminal of the audio amplification circuit for receipt of an audio input signal and operable to provide a first amplified audio signal with a first signal amplification;a second preamplifier having an input operatively coupled to an input terminal of the audio amplification circuit for receipt of an audio input signal and operable to provide a second amplified audio signal with a second signal amplification, smaller than the first signal amplification;a switch having a first input operatively coupled to the first preamplifier, a second input operatively coupled to the second preamplifier, and an output;an analogue-to-digital converter operatively coupled to the output of the switch and operable to provide a digital audio signal;a signal selection circuit operable to control the switch to selectively provide one of the first and second amplified audio signals on the output of the switch.2. An audio amplification circuit ...

Method and apparatus to shape sound

The invention relates to sonic steerable antennae and their use to achieve a variety of effects. The invention comprises a method and apparatus for taking an input signal, replicating it a number of times and modifying each of the replicas before routing them to respective output transducers such that a desired sound field is created. This sound field may comprise a directed beam, focus beam or a simulated origin. Further, “anti-sound” may be directed so as to create nulls (quiet spots) in an already existing sound field. The input signal replicas may also be modified in way which changes their amplitude or they may be filtered to provide the desired delaying. Reflective or resonant surfaces may be used to achieve a surround sound effect, a microphone may be located in front of an array of loudspeakers, beams of light may be used to identify the present focal position, a limiting device may be used to ensure that clipping or distortion is reduced when more than one input signal is output by the same device and the concept of beam directivity may be used to achieve input nulls or beams in a microphone made up of an array of input transducers. Further, sound field shaping information may be associated with an audio signal to be broadcast.

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 ...

Near-field null and beamforming

Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker.

Audio playing device having earphone jack with rotation detector

An audio playing device may operate in a first mode or a second mode. The audio playing device includes a speaker, an earphone jack receiving an earphone plug of an earphone, a sound output driver, a rotation angle detector, and a processor. The sound output driver only drives the speaker when the audio playing device operates in the first mode and only drives the earphone when the audio playing device operates in the second mode. The rotation angle detector generates a first mode change signal when the earphone plug is rotated in the earphone jack along a first direction over a predetermined angle. The processor changes operation of the audio playing device between the first mode and the second mode when the processor receives the first mode change signal from the rotation angle detector.

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.

Analog Signal Transfer System, Variable Compressor, and Variable Expander

An analog signal transfer system includes a transmission apparatus including a variable compressor that variably compresses input signals exponentially according to the amplitudes of the input signals; and a reception apparatus including a variable expander that variably expands the compressed signals exponentially according to the amplitudes of the compressed signals. 1. An analog signal transfer system comprising:a transmission apparatus including a variable compressor that variably compresses input signals exponentially according to amplitudes of the input signals and generates compressed signals; anda reception apparatus including a variable expander that variably expands the compressed signals exponentially according to amplitudes of the compressed signals and generates expanded signals.3. The analog signal transfer system according to claim 1 , wherein the variable compressor variably compresses the input signals by feedback control.4. The analog signal transfer system according to claim 1 , wherein the variable expander variably expands the compressed signals by feedforward control.5. The analog signal transfer system according to claim 1 , whereinthe variable compressor variably compresses the input signals by continuously varying a compression ratio from 1:1 to 1:N exponentially according to the amplitudes of the input signals, andthe variable expander variably expands the compressed signals by continuously varying an expansion ratio from 1:N to 1:1 exponentially according to the amplitudes of the compressed signals.6. The analog signal transfer system according to claim 1 , whereinthe variable compressor attenuates the amplitude of the input signals having amplitudes larger than a predetermined value and generates the compressed signals,the variable expander amplifies the amplitude of the compressed signals having amplitudes larger than a predetermined value and generates the expanded signals.7. A variable compressor that variably compresses input signals ...

SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD

A signal processing apparatus includes an adder that acquires a plurality of input signals from a plurality of microphones and calculates an added value obtained by adding the input signals together, a subtracter that acquires a plurality of input signals from the plurality of microphones and calculates a subtracted value obtained by subtracting one input signal from the other input signal, and a determination unit that determines whether noise is included in the input signals based on the added value and the subtracted value. 1. A signal processing apparatus comprising:an adder that acquires a plurality of input signals from a plurality of microphones and calculates an added value obtained by adding the input signals together;a subtracter that acquires a plurality of input signals from the plurality of microphones and calculates a subtracted value obtained by subtracting one input signal from the other input signal; anda determination unit that determines whether noise is included in the input signals based on the added value and the subtracted value.2. The signal processing apparatus according to claim 1 , further comprising:a gain calculator that calculates a gain based on the added value and the subtracted value; anda noise suppression unit that suppresses noise of the input signals based on the gain calculated by the gain calculator.3. The signal processing apparatus according to claim 1 , wherein the determination unit determines that noise is included in the input signals when a difference between the added value and the subtracted value is smaller than a threshold value.4. The signal processing apparatus according to claim 1 , further comprisinga ratio calculator that calculates a ratio of the subtracted value to the added value,wherein the determination unit determines that noise is included in the input signals when a value of the ratio calculated by the ratio calculator is greater than or equal to a threshold value.5. The signal processing apparatus ...

MICROPHONE UNIT AND ELECTRONIC APPARATUS

In this microphone unit, a first sound hole and a second sound hole are provided to extend toward a surface of an electronic apparatus internally mounted with a differential vibrating portion, intersecting with a main surface of the electronic apparatus, and an end portion of the first sound hole closer to the surface intersecting with the main surface of the electronic apparatus and an end portion of the second sound hole closer to the surface intersecting with the main surface of the electronic apparatus are so arranged that the vertical distances of the end portions from the main surface of the electronic apparatus are different from each other. 1. A microphone unit comprising a differential vibrating portion detecting a sound wave on the basis of a difference between a sound pressure arriving through a first sound hole and a sound pressure arriving through a second sound hole , whereinthe first sound hole and the second sound hole are provided to extend toward a surface of an electronic apparatus internally mounted with the differential vibrating portion, intersecting with a main surface of the electronic apparatus, andan end portion of the first sound hole closer to the surface intersecting with the main surface of the electronic apparatus and an end portion of the second sound hole closer to the surface intersecting with the main surface of the electronic apparatus are so arranged that vertical distances of the end portions from the main surface of the electronic apparatus are different from each other.2. The microphone unit according to claim 1 , whereinthe first sound hole and the second sound hole are provided to extend toward the common surface intersecting with the main surface of the electronic apparatus.3. The microphone unit according to claim 1 , whereinthe end portion of the first sound hole and the end portion of the second sound hole are spaced from each other on the same axis line substantially orthogonal to the main surface of the electronic ...

ELECTRO-ACOUSTIC TRANSDUCER AND METHOD OF MANUFACTURING THE SAME

An electro-acoustic transducer includes an insulative flexible substrate, a base, and a magnetic field generator. The base includes a cavity and a magnetic portion disposed below the cavity. The insulative flexible substrate is configured to cover the cavity. The magnetic field generator can be disposed on the insulative flexible substrate and corresponds to the cavity. The magnetic field generator can produce a magnetic field and a reverse magnetic field to cause the magnetic field generator and the magnetic portion of the base to attract and repel each other, thereby vibrating the insulative flexible substrate. 1. An electro-acoustic transducer comprising:an insulative flexible substrate;a base comprising a cavity covered by the insulative flexible substrate and a magnetic portion disposed below the cavity; anda magnetic field generator disposed on the insulative flexible substrate and corresponding to the cavity;wherein the magnetic field generator is configured to produce a magnetic field and a reverse magnetic field to cause the magnetic field generator and the magnetic portion of the base to attract and repel each other, thereby vibrating the insulative flexible substrate.2. The electro-acoustic transducer of claim 1 , wherein the magnetic field generator comprises a planar coil.3. The electro-acoustic transducer of claim 2 , wherein the magnetic field generator comprises a conductive adhesive.4. The electro-acoustic transducer of claim 1 , wherein the magnetic portion comprises a polymer and magnetic powder.5. The electro-acoustic transducer of claim 1 , wherein the base comprises a polymer and magnetic powder.6. The electro-acoustic transducer of claim 1 , further comprising a signal processor coupled with the magnetic field generator claim 1 , wherein the signal processor is configured to drive the magnetic field generator to produce the magnetic field and the reverse magnetic field.7. The electro-acoustic transducer of claim 6 , further comprising a ...

System And Method For Automatically Remixing Digital Music

Systems and methods augment a target media with a plurality of source media. The target media and source media are processed to form time frequency distributions (TFDs). Target features are extracted from the associated TFD and source features are extracted from each of the associated source TFDs. The target features are segmented into temporal portions that are compared with each of the plurality of source features to determine one or more matched source features having nearest matches to the target feature segments. Portions of the source media associated with the matched source features are mixed with the target media to form an augmented target media, wherein the mixing is based upon a probabilistic mixing algorithm that uses a distance between the matched target feature and source features to define an amplitude of each portion of the source media. 1. A method for generating a transformed digital media , including:decoding a target media file and one or more source media files into a target sample set and a plurality of source sample sets;processing the target sample set to form a target time frequency distribution (TFD);processing the source sample sets to form at least one source TFD;extracting musical features from the target TFD to generate target feature extraction (FE) data containing one or more separate target features;extracting musical features from the source TFD to generate source FE data containing one or more separate source features, and assigning each of the source features a media ID for identifying the associated source;segmenting the target FE data into temporal portions to form a plurality of target feature segments for each target feature, each of the target feature segments being time aligned to a corresponding target feature;comparing each of the target feature segments with a plurality of the source features and generating a triple for each substantial match between the target feature segment and one of the source features, the triple ...

PROGRAMMABLE MICROPHONE

A semiconductor die with an integrated electronic circuit, configured so as to be mounted in a housing with a capacitive transducer e.g. a microphone. A first circuit is configured to receive an input signal from the transducer at an input node and to provide an output signal at a pad of the semiconductor die. The integrated electronic circuit comprises an active switch device with a control input, coupled to a pad of the semiconductor die, to operatively engage or disengage a second circuit interconnected with the first circuit so as to operate the integrated electronic circuit in a mode selected by the control input. That is, a programmable or controllable transducer. The second circuit is interconnected with the first circuit so as to be separate from the input node. Thereby less noise is induced, a more precise control of the circuit is obtainable and more advanced control options are possible. 122-. (canceled)23. A semiconductor die with an integrated electronic circuit , for mounting in a microphone capsule of a capacitor microphone , the integrated electronic circuit comprising:a signal conditioner including a first circuit and a second circuit, the second circuit connected to the first circuit, the first circuit configured to receive an input signal at an input node from a capacitive transducer via a first terminal of the semiconductor die and to provide an output signal at an output terminal of the microphone capsule via a second terminal of the semiconductor die; anda mode detector coupled to at least one of a clock signal input or a power input of the semiconductor die, the mode detector configured to select a mode of the signal conditioner by engaging or disengaging the second circuit by a mode select signal.24. The semiconductor die of claim 23 , wherein the mode detector is coupled to the clock signal input to detect into which one of a plurality of predefined frequency ranges the clock signal falls and selects the mode of the signal conditioner based ...

Multi-Coaxial Transducers and Methods

Coaxial transducers, some of which include a first assembly and assembly, each of which includes a magnet and a coil. 119.-. (canceled)20. A transducer comprising:two coaxially-arranged assemblies that are coupled together, each assembly including a magnet and a coil, where one assembly at least partially overlaps the other assembly; anda housing to which one of the assemblies is coupled through at least one suspension element.21. The transducer of claim 20 , further comprising N assemblies claim 20 , where N is greater than or equal to 3 and each assembly includes a magnet and a coil.22. The transducer of claim 20 , where each assembly includes a coil former coupled to the coil of that assembly and to the housing.23. The transducer of claim 22 , where at least one of the coil formers comprises one or more ventilation holes.24. The transducer of claim 20 , where each assembly includes a flux focuser coupled to the magnet of that assembly.2526.-. (canceled)27. The transducer of claim 20 , where the at least one suspension element comprises a spring.28. The transducer of claim 22 , where the external housing comprises an output base to which the coil formers are attached.29. The transducer of claim 28 , wherein the external housing further comprises a top that is coupled to the output base and to one of the assemblies by the at least one suspension element.30. A transducer comprising:a housing;a first magnet positioned inside the housing;a first coil positioned around at least a portion of the first magnet, the first coil being coupled to the housing, the first coil having a first outer perimeter;a second magnet coupled to the first magnet; anda second coil positioned around at least a portion of the second magnet, the second coil being coupled to the housing in substantially coaxial alignment with the first coil and having a second outer perimeter that is less than the first outer perimeter;where the first magnet is coupled to the housing and to the second magnet ...

Magnetic circuit for speaker and speaker using same

A bond magnet according to the present invention is anisotropic, and a magnetic field is oriented from a lowest edge of side surface of the bond magnet toward an upper section of side surface of the bond magnet. Further, the bond magnet is configured such that the upper section of side surface of the bond magnet is disposed closer to a yoke side surface of a yoke than the lower side surface of the bond magnet is, and the upper section of side surface of the bond magnet faces against the yoke side surface. With this, a magnetic gap is provided between the upper section of side surface of the bond magnet and the yoke side surface. Thus, it is possible to improve a magnetism characteristic of the bond magnet, and to achieve both of magnetic efficiency and productivity of a magnetic circuit for a speaker.

Sine wave generating device, digital signal processor and audio output device

A sine wave generating device includes at least one adder configured to add two input signals thereof, at least one delay unit configured to delay an input signal thereof by one sample time and at least one multiplier configured to receive the delayed signal from the at least one delay unit, multiply the delayed signal by a coefficient and output the multiplied signal to provide to the at least one adder. The coefficient is arbitrarily set from outside the sine wave generating device.

HEAD-RELATED TRANSFER FUNCTION (HRTF) SELECTION OR ADAPTATION BASED ON HEAD SIZE

A method includes determining a user's head size, forwarding information associated with the user's head size to a processing device. The method also includes identifying, by the processing device, a head-related transfer function (HRTF) associated with the user's head size or modifying, by the processing device, a HRTF based on the user's head size. The method further includes applying the identified HRTF or modified HRTF to audio signals to produce output signals, and forwarding the output signals to first and second speakers. 1. A method comprising:determining a user's head size;forwarding information associated with the user's head size to a processing device; ["identifying, by the processing device, a head-related transfer function (HRTF) associated with the user's head size, or", "modifying, by the processing device, a HRTF based on the user's head size;"], 'at least one ofapplying the identified HRTF or modified HRTF to audio signals to produce output signals; andforwarding the output signals to first and second speakers.2. The method of claim 1 , wherein the at least one of identifying or modifying comprises identifying a HRTF associated with the user's head size claim 1 , wherein the identifying comprises:accessing a memory storing a plurality of HRTFs, andidentifying a first one of the plurality of HRTFs corresponding to the user's head size.3. The method of claim 2 , wherein the memory is configured to at least one of:store HRTFs corresponding to a small head size, a medium head size and a large head size,store HRTFs corresponding to a plurality of different head circumferences, orstore HRTFs corresponding to a plurality of different head diameters.4. The method of claim 1 , further comprising:determining a second user's head size;forwarding information associated with the second user's head size to the processing device; andaccessing the memory to determine whether one of the plurality of HRTFs corresponds to the second user's head size.5. The method of ...

System for wireless switching and controlling of audio signals as related to musical audio applications such as audio effects, effects pedals, pedalboards, mixers, and studio equipment

A system for wireless control of audio signal switching. The system includes a wireless transmitter having a physically actuated switch, and a main unit. The main unit includes a wireless receiver and one or more relays which may switch the path of the audio signal or provide remote control of external devices, in response to input at the physically actuated switch.

AUDIO SYSTEM WITH SYNTHESIZED POSITIVE IMPEDANCE

An electrical apparatus to sense current through a load includes a first input terminal having a first input voltage relative to a reference, a second input terminal having a second input voltage relative to the reference, a first load terminal of the load having a first load voltage relative to the reference, a second load terminal of the load having a second load voltage relative to the reference, a first current sensing element connected between the first input terminal and the first load terminal and a second current sensing element connected between the second input terminal and the second load terminal. A first sense voltage is determined by a relationship between the first input voltage and the second load voltage and a second sense voltage is determined by a relationship between the second input voltage and the first load voltage. 125-. (canceled)26. An electrical apparatus to sense current through a load comprising:a first input terminal having a first input voltage relative to a reference;a second input terminal having a second input voltage relative to the reference;a first load terminal of the load having a first load voltage relative to the reference;a second load terminal of the load having a second load voltage relative to the reference;a first current sensing element connected between the first input terminal and the first load terminal; anda second current sensing element connected between the second input terminal and the second load terminal, wherein a first sense voltage is determined by a relationship between the first input voltage and the second load voltage and a second sense voltage is determined by a relationship between the second input voltage and the first load voltage.27. The apparatus of wherein the reference is selected from the group consisting of a circuit common claim 26 , a circuit ground claim 26 , and an earth connection.28. The apparatus of wherein the first current sensing element is a resistive element.29. The apparatus of ...

Apparatus, Method and Computer Program for Controlling an Acoustic Signal

An apparatus comprising: at least one filter configured to filter an electrical input signal and provide a filtered electrical input signal to at least one speaker module configured to convert the filtered electrical input signal to an acoustic signal; and a detector configured to receive the filtered electrical input signal as a first input and an electrical output signal provided by at least one microphone as a second input; wherein the detector is configured to determine at least one difference between the electrical output signal provided by the at least one microphone and the filtered electrical input signal provided to said speaker module and, in response to the at least one difference provide a control signal to the filter to control the filter. 121-. (canceled)22. An apparatus comprising:at least one filter configured to filter an electrical input signal and provide a filtered electrical input signal to at least one speaker module, said speaker module configured to convert the filtered electrical input signal to an acoustic signal; anda detector configured to receive the filtered electrical input signal as an input and an electrical output signal provided by at least one microphone; whereinthe detector is configured to determine at least one difference between the electrical output signal provided by the at least one microphone and the filtered electrical input signal provided to said speaker module and, in response to the at least one difference provide a control signal to the at least one filter to control the filter.23. The apparatus as claimed in claim 22 , wherein the at least one difference comprises at least one of:a difference between a frequency response of the filtered electrical input signal and a frequency response of the electrical output signal provided by the at least one microphone;a difference between a signal level of the filtered electrical input signal and a signal level of the electrical output signal provided by the at least one ...

AUDIO SIGNAL PROCESSING SYSTEM AND RECORDING METHOD

A processing apparatus communicatively connected with a mixer apparatus and having an audio recording function is capable of recording in real time audio signals of one or more channels output from the mixer apparatus. When a snapshot change is to be made for collectively changing a state of a set of signal-processing setting data, the mixer apparatus transmits, to the processing apparatus, a command for setting a given parameter. Upon receipt of the command, the processing sets and records the given parameter, instructed by the received command, into a project that is recording in real time the audio signals of one or more channels. 1. An audio signal processing system comprising:a mixer apparatus; anda processing apparatus communicatively connected with said mixer apparatus and having an audio recording function, said processing apparatus being configured to be capable of recording in real time audio signals of one or more channels output from said mixer apparatus,wherein said mixer apparatus is configured to transmit, to said processing apparatus, a command for setting a given parameter when a snapshot change is to be made for collectively changing a state of a set of signal-processing setting data, andsaid processing apparatus is configured to, upon receipt of the command, set the given parameter, instructed by the received command, into a project that is recording in real time the audio signals of one or more channels.2. The audio signal processing system as claimed in claim 1 , wherein said mixer apparatus is configured to be capable of making claim 1 , based on user's operation claim 1 , a setting as to whether or not the command should be transmitted claim 1 , when a snapshot change is to be made for collectively changing a state of a set of signal-processing setting data.3. The audio signal processing system as claimed in claim 2 , wherein said mixer apparatus is capable of claim 2 , for each of one or more types of parameters claim 2 , transmitting the ...

Equalization of an Audio Signal

A method of processing an input audio signal, the method comprising forming a plurality of output audio signals from the input audio signal, wherein each output audio signal is formed by performing respective processing on the input audio signal, wherein for a first output audio signal there is a target audio equalization operation comprising a target filter twice, wherein for the first output 10 audio signal, the respective processing comprises a first audio equalization operation, the first audio equalization operation being the target audio equalization operation modified to compensate for phase shifts that correspond to zeros of the transfer function of the target audio equalization operation, wherein for each output audio signal other than the first output audio signal, the respective 15 processing comprises a compensation filter that compensates for phase shifts that correspond to poles of the transfer function of the target audio equalization operation. 20 1. A method of processing an input audio signal , the method comprising:forming a plurality of output audio signals from the input audio signal, wherein each output audio signal is formed by performing respective processing on the input audio signal, wherein for a first output audio signal there is a target audio equalization operation comprising a target filter twice;wherein for the first output audio signal, the respective processing comprises a first audio equalization operation, the first audio equalization operation being the target audio equalization operation modified to compensate for phase shifts that correspond to zeros of the transfer function of the target audio equalization operation;wherein for each output audio signal other than the first output audio signal, the respective processing comprises a compensation filter that compensates for phase shifts that correspond to poles of the transfer function of the target audio equalization operation.2. A method of determining a configuration for audio ...

MULTI-DIRECTIONAL AND OMNIDIRECTIONAL HYBRID MICROPHONE FOR HEARING ASSISTANCE DEVICES

Disclosed herein, among other things, are methods and apparatus for an directional microphone arrays for hearing assistance devices. In various embodiments, the present subject matter provides a microphone array system for receiving sounds including a first directional microphone, a second directional microphone and an omnidirectional microphone. The first directional microphone has a first directional axis in a first direction, and the second directional microphone has a second directional axis that is collinear with the first direction and pointing in the same direction as the first direction. The omnidirectional microphone has a sound sampling position that is a disposed between the first directional microphone and the second directional microphone, and the omnidirectional microphone sound sampling position is on or about the first directional axis. Weighted outputs of the first directional microphone, second directional microphone, and omnidirectional microphone are processed to provide a second order directional microphone system, according to various embodiments. 1. A microphone array system for receiving sounds , comprising:a first directional microphone having a first directional axis in a first direction;a second directional microphone having a second directional axis that is collinear with the first direction and pointing in the same direction as the first direction;an omnidirectional microphone having a sound sampling position that is a disposed between the first directional microphone and the second directional microphone;wherein the omnidirectional microphone sound sampling position is on or about the first directional axis and wherein weighted outputs of the first directional microphone, second directional microphone, and omnidirectional microphone are processed to provide a second order directional microphone system.2. The system of claim 1 , wherein the first directional microphone claim 1 , the second directional microphone and the omnidirectional ...

AUDIO SIGNAL PROCESSING SYSTEM

An audio signal processing system for recording a sound inputted into a channel of a mixer which receives a plurality of audio signals, processes the audio signals in a plurality of channels and outputs the processed signals, and reproducing the recorded audio signal for adjusting parameters of the mixer is provided. The audio signal processing system records, when record is instructed a channel is selected, an audio signal inputted to the selected channel with relation information indicating the selected channel, and reproduces, when reproduction is instructed and a channel is selected, an audio signal which has been stored with relation information indicating the selected channel. 1. An audio signal processing system for recording a sound inputted into a channel of a mixer which receives a plurality of audio signals , processes the audio signals in a plurality of channels and outputs the processed signals , and reproducing the recorded sound for adjusting parameters of the mixer , comprising:a record instructor that instructs to record;a reproduction instructor that instructs to reproduce;a channel selector that selects a channel as a target;a recorder that, when the record instructor instructs to record and the channel selector selects a first channel, records an audio signal inputted to the selected first channel with relation information indicating the selected first channel;a reproducer that, when the reproduction instructor instructs to reproduce and the channel selector selects a second channel, reproduces an audio signal which has been stored with relation information indicating the selected second channel, and sends the reproduced audio signal to the selected second channel.2. An audio signal processing system according to claim 1 ,wherein the recorder records audio signals of a plurality of takes regarding one channel with take information indicating the take of the respective recorded audio signals.3. An audio signal processing system according to claim ...

LOW POWER MICROPHONE CIRCUITS FOR VEHICLES

A low power microphone circuit for a vehicle is provided that includes: at least one microphone transducer; a digital signal processor for receiving output signals from the at least one microphone transducer and for generating a digitally processed audio signal; an output amplifier for amplifying the audio signal from the digital signal processor and modulating an input voltage with the audio signal; and a DC power supply for supplying power to the digital signal processor. The output amplifier and the DC power supply may be electrically coupled in series. The DC power supply and the output amplifier may be powered by the input current, where the input current is no greater than about 6 mA. 1. A low power microphone circuit for a vehicle , comprising:at least one microphone transducer;a digital signal processor for receiving output signals from said at least one microphone transducer and for generating a digitally processed audio signal;an output amplifier for amplifying the audio signal from said digital signal processor and modulating an input voltage with the audio signal; anda DC power supply for supplying power to said digital signal processor,wherein said output amplifier and said DC power supply are electrically coupled in series.2. The low power microphone circuit of and further including a terminal for connection to a vehicle power source providing the input voltage and an input current claim 1 , wherein said output amplifier coupled between said terminal and said DC power supply and wherein said DC power supply is coupled between said output amplifier and ground.3. The low power microphone circuit of claim 2 , wherein the input current is no greater than about 6 mA.4. The low power microphone circuit of claim 1 , wherein said DC power supply is a shunt regulator.5. The low power microphone circuit of claim 1 , wherein said DC power supply is a simulated inductor circuit.6. The low power microphone circuit of claim 1 , wherein said DC power supply provides ...

ADAPTIVE GAIN CONTROL FOR DIGITAL AUDIO SAMPLES IN A MEDIA STREAM

An adaptive gain control system and related operating method for digital audio samples is provided. The method is suitable for use with a digital media encoding system that transmits encoded media streams to a remotely-located presentation device such as a media player. The method begins by initializing the processing of a media stream. Then, the method adjusts the gain of a first set of digital audio samples in the media stream using a fast gain adaptation scheme, resulting in a first group of gain-adjusted digital audio samples having normalized volume during presentation. The method continues by adjusting the gain of a second set of digital audio samples in the media stream using a steady state gain adaptation scheme that is different than the fast gain adaptation scheme, resulting in a second group of gain-adjusted digital audio samples having normalized volume during presentation. 1. A computer program product , tangibly embodied in a computer-readable medium , the computer program product being operable to cause a digital media processing device to perform a method comprising:initializing processing of a media stream;adjusting gain of a first set of digital audio samples in the media stream using a fast gain adaptation scheme, resulting in a first group of gain-adjusted digital audio samples having normalized volume during presentation;transmitting the first group of gain-adjusted digital audio samples to a remotely-located media player to generate sound based on the first group of gain-adjusted digital audio samples;thereafter, adjusting gain of a second set of digital audio samples in the media stream using a steady state gain adaptation scheme that is different than the fast gain adaptation scheme, resulting in a second group of gain-adjusted digital audio samples having normalized volume during presentation, wherein the second set of digital audio samples follow the first set of digital audio samples, and wherein the second group of gain-adjusted digital ...

Method and apparatus to automatically control audio volume

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

Microphone apparatus and method for removing unwanted sounds

An apparatus comprises a first transducer configured to detect sound and generate a first signal based on the detected sound. The apparatus also comprises a second transducer configured to detect vibration and/or sound and generate a second signal based on the detected vibrations and/or sound. The second transducer is less acoustically responsive than the first transducer. The apparatus comprises an interface configured to send the first and second signals to a processor configured to modify the first signal on the basis of the second signal.

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.

MULTIPLE MICROPHONE SWITCHING AND CONFIGURATION

A mobile communications device contains at least two microphones. One microphone is designated by a selector to provide a voice dominant signal and another microphone is designated to provide a noise or echo dominant signal, for a call or a recording. The selector communicates the designations to a switch that routes the selected microphone signals to the inputs of a processor for voice signal enhancement. The selected voice dominant signal is then enhanced by suppressing ambient noise or canceling echo therein, based on the selected noise or echo dominant signal. The designation of microphones may change at any instant during the call or recording depending on various factors, e.g. based on the quality of the microphone signals. Other embodiments are also described. 1a mobile communications handheld device housing having integrated thereina plurality of microphones; andan audio processor to provide an uplink voice signal for a call by enhancing a voice dominant input signal using an echo or noise dominant input signal, the audio processor to run a switching process that monitors audio picked up by the microphones and compares signals from the microphones with each other and on that basis designates one of the microphones to provide the voice dominant input signal and another one of the microphones to provide the echo or noise dominant input signal, wherein the switching process is to dynamically change said designation of the microphones during the call based on a current call handling mode of the apparatus, being one of handset mode and speakerphone mode.. A mobile communications apparatus comprising: RELATED MATTERSThis application is a continuation of U.S. patent application Ser. No. 12/242,517, filed Sep. 30, 2008, entitled “Multiple Microphone Switching and Configuration” (to issue as U.S. Pat. No. 8,401,178 on Mar. 19, 2013).An embodiment of the invention is generally related to mobile devices that have multiple microphones for enhancing an audio signal. ...

Noise cancellation system

An earphone comprises an earphone body, containing a speaker, and a projection, extending from a first surface of the earphone body, for location in the entrance to the user's ear canal. The earphone body comprises a sound outlet in the first surface, for allowing sounds generated by the speaker to leave the earphone body. The projection extends from the first surface of the earphone body, adjacent to the sound outlet, and contains a sound inlet port, connected to a microphone for detecting sounds entering the ear canal. A noise cancellation system includes noise cancellation circuitry, for applying a frequency dependent filter characteristic and applying a gain to an input signal representing ambient noise, at least one of the frequency dependent filter characteristic and the gain being adaptive. The earphone then has an ambient noise microphone, and an error microphone connected to the sound inlet port.

HEARING APPARATUS HAVING AN ADAPTIVE FILTER AND METHOD FOR FILTERING AN AUDIO SIGNAL

A hearing apparatus that processes an audio signal by adaptive filtering operates efficiently. The hearing apparatus has a first stage transformation unit and at least two second stage transformation units. Each of the second stage transformation units splits the signal of one of the first stage channels through transformation onto sub-channels of the channel. A filter unit in each channel that has a second stage transformation unit filters the signal of the channel depending on weighting factors that are determined in each case for the sub-channels. The filter units are configured to exchange weighting factors among themselves, which means that only some of the weighting factors actually need to be calculated. 1. A method for the adaptive filtering of an audio signal , the method which comprises:transforming the audio signal by a first transformation stage and thereby generating a sub-band signal in each of a plurality of channels; and transforming the sub-band signal by a second transformation stage and splitting onto a plurality of sub-channels of the channel; and', 'in each sub-channel, defining a weighting factor for the signal of the sub-channel by the filter unit of the channel;, 'in at least some of the plurality of channels, adaptively filtering the respective sub-band signal by a filter unit of the channel, and thereby in the channelwherein, with regard to at least one of the filter units, the weighting factor of at least one sub-channel is defined by calculation depending on the signal of the sub-channel itself and the calculated weighting factor is transferred by the filter unit to at least one other filter unit, and with regard to the at least one other filter unit, the weighting factor of at least one sub-channel is defined on the basis of the received, calculated weighting factor.2. The method according to claim 1 , which comprises transferring the calculated weighting factor of a first sub-channel to a sub-channel that overlaps spectrally with the ...

Mobile Microphone System and Method

A method centrally controls communication at a single venue. Information, at least at the venue, is provided to attendees at the venue for transmitting electronic signals to the central communication point. At least one electronic communication transmission from an attendee who has access to the central communication point is enabled to have at least part of the transmission broadcast at the venue or allow open communication from the attendee to be broadcast at the venue. The personal mobile communications device of an attendee is used as a microphone over wireless telephone networks or local wireless communication networks. 1. For a central communications point and a system communicating sound from a central speaker to an audience , a method of wirelessly transmitting sound from an audience member back to the audience through an amplified audio system , comprising:enabling a personal mobile communications device used by the audience member, the personal mobile communications device nominally communicating audio signals over a first wireless communications channel, to communicate audio signals corresponding to the sound from the audience member over a local wireless communications channel;receiving the audio signals from the personal mobile communications device over the local wireless communications channel through the central communication point; andsending the audio signals from the central communications point to the audio system;whereby sound corresponding to the audio signals is broadcast to the audience by the amplified audio system without significant delay.2. The method of wherein the local wireless communications channel uses RF (Radio Frequency) signals.3. The method of wherein the RF signals comprise WiFi signals.4. The method of wherein the RF signals comprise FM (Frequency Modulation) signals.5. The method of wherein the local wireless communications channel uses IR (Infrared) signals.6. The method of wherein the enabling step comprises:providing the ...

MULTI-BEAM SOUND SYSTEM

A multi-beam sound system includes a fixed beamforming section which steers the input signal inputted from the microphone array to an intended direction, a blocking matrix which receives the input signal and acquires a noise reference signal from the input signal, a variable beamforming section which acquires an adaptive noise signal from the noise reference signal outputted from the blocking matrix, and a generalized sidelobe canceller (GSC) which includes canceling means for outputting an object signal from the input signal outputted from the fixed beamforming section by removing the adaptive noise signal from the input signal. The fixed beamforming section steers the input signal in at least two directions. 1. A multi-beam sound system comprising:a microphone array disposed at a predetermined position inside a vehicle, the microphone array comprising a plurality of microphones, and receiving an input signal; andan adaptive beamformer which forms beams of the microphone array,wherein the adaptive beamformer forms at least two beams of the microphone array in different directions.2. The multi-beam sound system according to claim 1 , wherein the beamformer comprising:a fixed beamforming section which steers the input signal inputted from the microphone array to an intended direction;a blocking matrix which receives the input signal and acquires a noise reference signal from the input signal;a variable beamforming section which acquires an adaptive noise signal from the noise reference signal outputted from the blocking matrix; anda generalized sidelobe canceller (GSC) which includes canceling means for outputting an object signal from the input signal outputted from the fixed beamforming section by removing the adaptive noise signal from the input signal,wherein the fixed beamforming section steers the input signal in at least two directions.4. The multi-beam sound system according to claim 2 , wherein the variable beamforming section acquires the adaptive noise ...

Headphone, headphone stand and headphone system

A wireless headphone system 102 includes a headphone 144 and a headphone stand 104 . The headphone 144 has two ear cups 150 a and 150 b . The ear cup 150 a incorporates a receiving coil L 3 and a capacitor C 3 . The ear cup 150 b incorporates a charge controller 138 and a secondary battery 112. The headphone stand 104 incorporates a feeding coil L 2. When the headphone 144 is set in the headphone stand 104, AC power is supplied from the feeding coil L 2 to the receiving coil L 3, and the secondary battery 112 of the headphone 144 is charged with the AC power.

ELECTRONIC DEVICE, ACCESSORY, AND METHOD FOR DETECTING AN ACCESSORY

Electronic device, accessory, and method for detecting an accessory are provided. The electronic device includes a connector, a control unit, and an audio processing unit. The connector is configured to connect to an accessory and has an audio contact. The control unit has first and second terminals, which are coupled to the audio contact. The control unit is configured to send a command, which is represented by a first serial digital signal, to the accessory through the first terminal and audio contact when the accessory is electrically connected to the connector, and to receive accessory data, which is represented by a second serial digital signal and generated by the accessory in response to the command, through the second terminal and audio contact. The audio processing unit is coupled between the audio contact and control unit, and configured to receive an audio signal from the accessory through the audio contact. 1. An electronic device , comprising:a connector for connecting to an accessory, the connector having an audio contact;a control unit having a first terminal and a second terminal, which are coupled to the audio contact of the connector, wherein the control unit is configured to send a command, which is represented by a first serial digital signal, to the accessory through the first terminal and the audio contact when the accessory is electrically connected to the connector, and to receive accessory data, which is represented by a second serial digital signal and generated by the accessory in response to the command, through the second terminal and the audio contact; andan audio processing unit coupled between the audio contact and the control unit and configured to receive an audio signal from the accessory through the audio contact.2. The electronic device according to claim 1 , further comprising:an accessory data detection device coupled between the audio contact and the second terminal of the control unit and configured to receive the accessory ...

Wireless Microphone System With Frequency Matching Capability, And Method For Frequency Matching In A Wireless Microphone System

A wireless microphone system includes a receiver device, a wireless microphone device, a transmitting unit operable to transmit an identification code and a frequency matching signal, and a reading unit. The reading unit is operable to receive the identification code and the frequency matching signal, determine whether or not the identification code corresponds to the reading unit, and enable frequency matching between the receiver device and the wireless microphone device according to the frequency matching signal when the identification code is determined to correspond to the reading unit. 1. A wireless microphone system comprising:a receiver device;a wireless microphone device;a transmitting unit disposed in one of said receiver device and said wireless microphone device, and operable to transmit an identification code and a frequency matching signal; anda reading unit disposed in the other one of said receiver device and said wireless microphone device, and operable toreceive the identification code and the frequency matching signal transmitted by said transmitting unit,determine whether or not the identification code corresponds to said reading unit, andenable the other one of said receiver device and said wireless microphone device to perform frequency matching with said one of said receiver device and said wireless microphone device according to the frequency matching signal when the identification code is determined to correspond to said reading unit.2. The wireless microphone system as claimed in claim 1 , wherein said transmitting unit is disposed in said receiver device claim 1 , and said reading unit is disposed in said wireless microphone device.3. The wireless microphone system as claimed in claim 2 , wherein said reading unit is configured to change a microphone frequency of said wireless microphone device according to the frequency matching signal to perform frequency matching with said receiver device when the identification code is determined to ...

APPARATUS AND METHOD FOR DERIVING A DIRECTIONAL INFORMATION AND COMPUTER PROGRAM PRODUCT

An apparatus for deriving a directional information from a plurality of microphone signals or from a plurality of components of a microphone signal, wherein different effective microphone look directions are associated with the microphone signals or components, has a combiner configured to obtain a magnitude value from a microphone signal or a component of the microphone signal. The combiner is further configured to combine direction information items describing the effective microphone look directions, such that a direction information item describing a given effective microphone look direction is weighted in dependence on the magnitude value of the microphone signal, or of the component of the microphone signal, associated with the given effective microphone look direction, to derive the directional information. 1. Apparatus for deriving a directional information from a plurality of microphone signals or from a plurality of components of a microphone signal , wherein different effective microphone look directions are associated with the microphone signals or components , the apparatus comprising:a combiner configured to acquire a magnitude value from a microphone signal or a component of the microphone signal, and to combine direction information items describing the effective microphone look directions, such that a direction information item describing a given effective microphone look direction is weighted in dependence on the magnitude value of the microphone signal, or of the component of the microphone signal, associated with the given effective microphone look direction, to derive the directional information;wherein a direction information item describing a given effective microphone look direction is a vector pointing in the given effective microphone look direction;wherein the combiner is configured to derive the directional information d(k, n) for a given time frequency tile corresponding to a linear combination of the direction information items weighted ...

AUDIO MIXING SYSTEM

An audio mixing system comprises a plurality of input ports and output ports, and a plurality of input channels and output channels for controlling characteristic of input sound signals. Two or more input ports included in the plurality of input ports or two or more output ports included in the plurality of output ports are registered as a plurality of patch ports. By an instruction to collectively patch a port group, the input ports or the output ports registered as the plurality of patch ports are patched to input channels or output channels respectively by an input or an output patch. 1. An audio mixing system comprising:a plurality of input ports adapted to receive input sound signals;a plurality of input channels adapted to receive input sound signals from the plurality of input ports;an input patch portion adapted for selectively patching the plurality of input ports to the plurality of input channels;a mix bus adapted for mixing sound signals supplied from the plurality of input channels;a plurality of output channels adapted for inputting sound signal mixed by the mix bus;a plurality of output ports adapted to output sound signals;an output patch portion adapted for selectively patching the plurality of output ports to the plurality of output channels; anda port registration portion adapted for registering two or more input ports included in the plurality of input ports or two or more of output ports included in the plurality of output ports as a plurality of patch ports, whereinthe input patch portion or the output patch portion is provided with a group patch portion adapted for patching the plurality of patch ports registered by the port registration portion to two or more channels included in the plurality of input channels or two of more channels included in the plurality of output channels, respectively.2. The audio mixing system according to claim 1 , whereinthe port registration portion is capable of registering a plurality of port groups each formed ...

Processing Method for Earphone and User Equipment

A processing method for an earphone includes receiving a sense signal sent by a proximity sensor. A distance between an earphone and an ear corresponding to the earphone is obtained according to the sense signal and compared with a first preset threshold. Within a first preset time, if the distance is greater than the first preset threshold, an audio playing program is closed or an earphone channel of the earphone is closed. 1. A processing method for an earphone , the method comprising:obtaining a sense signal;obtaining a distance between an earphone and an ear corresponding to the earphone according to the sense signal;comparing the distance with a first preset threshold; andwithin a first preset time, closing an audio connection of the earphone if the distance is greater than the first preset threshold.2. The method according to claim 1 , wherein closing the audio connection comprises closing an audio playing program.3. The method according to claim 1 , wherein closing the audio connection comprises closing an earphone channel of the earphone.4. The method according to claim 1 , wherein the earphone comprises a first earphone and a second earphone claim 1 , the distance comprises a first distance between the first earphone and a first ear corresponding to the first earphone and a second distance between the second earphone and a second ear corresponding to the second earphone claim 1 , and the closing comprises:within the first preset time, if the first distance is greater than the first preset threshold and the second distance is less than or equal to the first preset threshold, closing an audio playing program or closing an earphone channel of the first earphone.5. The method according to claim 4 , further comprising storing audio information when the audio playing program is being closed.6. The method according to claim 5 , further comprising:within a second preset time, if the first distance is less than or equal to the first preset threshold, starting the ...

Support Apparatus for Microphone Diaphragm

A microphone includes a diaphragm assembly supported by a substrate. The diaphragm assembly includes at least one carrier, a diaphragm, and at least one spring coupling the diaphragm to the at least one carrier such that the diaphragm is spaced from the at least one carrier. An insulator (or separate insulators) between the substrate and the at least one carrier electrically isolates the diaphragm and the substrate.

Augmented reality (ar) audio with position and action triggered virtual sound effects

An augmented reality (AR) audio system for augmenting environment or ambient sound with sounds from a virtual speaker or sound source positioned at a location in the space surrounding an AR participant. The sound from the virtual speaker may be triggered by an action of the listener and/or by the location or relative orientation of the listener. The AR audio system includes stereo earphones receiving an augmented audio track from a control unit, and binaural microphones are provided to capture ambient sounds. The control unit operates to process trigger signals and retrieve one or more augmentation sounds. The control unit uses an AR audio mixer to combine the ambient sound from the microphones with the augmentation sounds to generate left and right ear augmented audio or binaural audio, which may be modified for acoustic effects of the environment including virtual objects in the environment or virtual characteristics of real objects.

Audio output apparatus

According to an embodiment, a switch of an audio output device selects any of an output of the buffer and an output of the zero data generation unit to output to the output port. The switch control unit controls the selection of the switch based on an instruction of an output control signal. The front address storage unit stores a setting value of a front address of the buffer. The offset value storage unit stores an offset value set in the buffer. The output address control unit outputs a value obtained by adding the offset value to the setting value of the front address as an output start address of the buffer.

SOUND GENERATION DEVICE, SOUND GENERATION METHOD AND STORAGE MEDIUM STORING SOUND GENERATION PROGRAM

Disclosed is a sound generation device including a receiver which receives a sound emission instructing signal in which time data is included and a difference calculator which calculates a difference between a timing indicated in the received time data and a timing when the sound emission instructing signal is received by the receiver, when the sound emission instructing signal in which the time data is included is received. The sound generation device further includes a histogram creator which creates a histogram on the basis of the calculated difference and a difference calculated previously, when the difference is calculated and a timing controller which controls a timing for supplying the received sound emission instructing signal to a sound emission unit on the basis of the calculated difference and a most frequent difference in the created histogram, when the difference is calculated. 1. A sound generation device , comprising:a receiver which receives a sound emission instructing signal which includes time data;a difference calculator which calculates a difference between a timing indicated in the received time data and a timing when the receiver receives the sound emission instructing signal, when the sound emission instructing signal is received;a histogram creator which creates a histogram on the basis of the calculated difference and a difference calculated previously, when the difference is calculated; anda timing controller which controls a timing for supplying the received sound emission instructing signal to a sound emission unit which is connected to the timing controller on the basis of the calculated difference and a most frequent difference in the created histogram, when the difference is calculated.2. The sound generation device according to further comprising a performance operator which is held by a performer claim 1 ,wherein a movement detection unit which detects a movement of a main body of the performance operator,', 'a sound emission ...

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.