СПОСОБ И УСТРОЙСТВО ДЛЯ ВОСПРОИЗВЕДЕНИЯ ТРЕХМЕРНОГО ЗВУКОВОГО СОПРОВОЖДЕНИЯ

Изобретение относится к средствам воспроизведения трехмерного звукового сопровождения. Технический результат заключается в увеличении точности локализации звукового сигнала. Получают информацию о глубине изображения, указывающую расстояние между, по меньшей мере, одним объектом в сигнале изображения и опорным местоположением. Получают информацию о глубине звука, указывающую расстояние между, по меньшей мере, одним звуковым объектом в звуковом сигнале и опорным местоположением, на основании информации о глубине изображения. Обеспечивают звуковую перспективу для, по меньшей мере, одного звукового объекта на основании информации о глубине звука. 3 н.з. и 18 з.п. ф-лы, 9 ил.

КЛАСТЕРИЗАЦИЯ АУДИООБЪЕКТОВ С СОХРАНЕНИЕМ МЕТАДАННЫХ

Изобретение относится к средствам для кластеризации аудиообъектов. Технический результат заключается в повышении эффективности кластеризации аудиообъектов. Классифицируют множество аудиообъектов в некоторое количество категорий на основании информации, предназначенной для сохранения в метаданных, связанных с множеством аудиообъектов. Назначают заданное количество кластеров категориям. Размещают аудиообъект в каждой из категорий по меньшей мере в один из кластеров в соответствии с упомянутым назначением. Назначение заданного количества кластеров категориям включает в себя этап, на котором определяют для каждой категории соответствующее количество кластеров, предназначенных для назначения этой категории, таким образом, что общее количество кластеров составляет в сумме заданное количество кластеров. 3 н. и 12 з.п. ф-лы, 4 ил.

ПАРАМЕТРИЧЕСКОЕ СОВМЕСТНОЕ КОДИРОВАНИЕ АУДИОИСТОЧНИКОВ

Изобретение касается кодирования множества сигналов аудиоисточников, которые должны быть переданы или сохранены с целью микширования сигналов для синтеза волнового поля, сигналов многоканального объемного или стереофонического аудио после декодирования сигналов источников. Предложенный способ обеспечивает эффективное кодирование при совместном кодировании сигналов источников по сравнению с их отдельным кодированием, даже когда между сигналами источников отсутствует избыточность. Это возможно с учетом статистических свойств сигналов источников, свойства метода кодирования и пространственного слуха. Сумма сигналов источников передается вместе со статистическими свойствами сигналов источников, которые главным образом определяют важные для восприятия пространственные признаки окончательно микшированных аудиоканалов. Сигналы источников восстанавливаются в приемнике так, что их статистические свойства приблизительно тождественны соответствующим свойствам первоначальных сигналов источников. Технический ...

ОЦЕНКА ПОЛОЖЕНИЙ ГРОМКОГОВОРИТЕЛЕЙ

Изобретение относится к средствам для определения оценок положений громкоговорителей в системах объемного звука. Технический результат заключается в повышении точности определения оценки положений громкоговорителей в системах объемного звука. Система содержит датчики движения (201, 203, 205), выполненные с возможностью определения данных движения для пользовательского переносного блока, где данные движения описывают перемещение пользовательского переносного блока. Пользовательское устройство ввода (207, 209) принимает пользовательские активизации, которые указывают, что по меньшей мере одно из текущего положения и ориентации пользовательского переносного блока ассоциируется с положением громкоговорителя, когда принимается пользовательская активизация. Пользовательская активизация может возникать в результате, например, нажатия на кнопку пользователем. Анализирующий процессор (211) затем формирует оценки положений громкоговорителей в ответ на данные движения и пользовательские активизации ...

УСТРОЙСТВО И СПОСОБ ДЛЯ УПРАВЛЕНИЯ ВХОДНЫМ АУДИОСИГНАЛОМ

Изобретение относится к области обработки аудиосигналов, в частности к области обработки пространственных аудиосигналов. Технический результат – повышение эффективности управления входным аудиосигналом в пределах пространственного аудиосценария. Устройство связано с пространственным источником аудио в пределах пространственного аудиосценария, причем пространственный источник аудио находится на определенном расстоянии от слушателя в пределах пространственного аудиосценария, причем устройство содержит возбудитель, выполненный с возможностью управления входным аудиосигналом для получения выходного аудиосигнала, и контроллер, выполненный с возможностью управления параметрами возбудителя для управления входным аудиосигналом на основании определенного расстояния. 2 н. и 11 з.п. ф-лы, 6 ил.

УСТРОЙСТВО И СПОСОБ ОБЕСПЕЧЕНИЯ МЕРЫ ПРОСТРАНСТВЕННОСТИ, СВЯЗАННОЙ С АУДИОПОТОКОМ

Изобретение относится к средствам для обеспечения меры пространственности, связанной с аудиопотоком. Технический результат заключается в повышении эффективности оценивания меры пространственности для аудиопотоков. Оценивают аудиоканалы аудиопотока для обеспечения меры пространственности, связанной с аудиопотоком следующим образом. Определяют меру сходства между первым набором аудиоканалов аудиопотока, подлежащего воспроизведению в одном или более первых пространственных слоях, и вторым набором аудиоканалов аудиопотока, подлежащего воспроизведению в одном или более вторых пространственных слоях. Определяют меру пространственности на основании меры сходства. Определяют порог маскирования на основании информации уровня первого набора аудиоканалов и сравнения порога маскирования с информацией уровня второго набора аудиоканалов. Увеличивают меру пространственности, когда сравнение указывает, что порог маскирования превышен информацией уровня второго набора аудиоканалов, и мера сходства указывает ...

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

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

УСТРОЙСТВО ДЛЯ ФОРМИРОВАНИЯ ВЫХОДНОГО ПРОСТРАНСТВЕННОГО МНОГОКАНАЛЬНОГО АУДИО СИГНАЛА

Изобретение относится к средствам формирования выходного пространственного многоканального аудиосигнала на основе входного аудиосигнала и входного параметра. Технический результат заключается в уменьшении вычислительных затрат процесса декодирования/рендеринга. Раскладывают входной аудио сигнал на основе входного параметра для получения первой компоненты сигнала и второй компоненты сигнала, отличающихся друг от друга. Выполняют рендеринг первой компоненты сигнала для получения первого представления сигнала с первым семантическим свойством и выполняют рендеринг второй компоненты сигнала для получения второго представления сигнала с вторым семантическим свойством, отличающимся от первого семантического свойства. Обрабатывают первое представление сигнала и второе представление сигнала для получения выходного пространственного многоканального звукового сигнала. 3 н. и 12 з.п. ф-лы, 8 ил.

Устройство и способ модифицирования вывода аудиосигнала устройства

Изобретение относится к области вычислительной техники для обработки аудио-сигнала. Технический результат заключается в повышении точности модификации первоначального исходящего аудио-сигнала. Технический результат достигается за счет определения уровня громкости изначального исходящего аудио-сигнала; сравнения уровня громкости с пороговым значением; на основе сравнения, контроля за воспроизведением первоначального исходящего аудио-сигнала путем выборочного выполнения: (i) в ответ на то, что уровень громкости ниже порогового значения, передачи идентичного аудио-сигнала каждому из динамиков для воспроизведения модифицированного исходящего аудио-сигнала, причем модифицированный исходящий аудио-сигнал представляет собой исходящий моно-сигнал; и (ii) в ответ на то, что уровень громкости превышает пороговое значение, передачи соответствующих аудио-сигналов динамикам для воспроизведения модифицированного исходящего аудио-сигнала, причем соответствующие аудио-сигналы отличаются друг от друга, ...

УСТРОЙСТВО И СПОСОБ ОБРАБОТКИ АУДИОСИГНАЛА

... 1. Устройство (100) обработки аудиосигнала, причем устройство (100) содержит ! блок (108) обнаружения, выполненный с возможностью обнаружения состояния расширения аудиосигнала; ! блок изменения расширения (107), выполненный с возможностью изменения характеристики расширения аудиосигнала, в зависимости от обнаруженного состояния расширения аудиосигнала. ! 2. Устройство (100) по п.1, в котором блок (108) обнаружения выполнен с возможностью обнаружения, был ли аудиосигнал предварительно расширен или сжат, или нет. ! 3. Устройство (100) по п.1, в котором блок (108) обнаружения выполнен с возможностью обнаружения параметра расширения, указывающего на степень, с которой аудиосигнал был предварительно расширен или сжат. ! 4. Устройство (100) по п.1, в котором блок (108) обнаружения выполнен с возможностью обнаружения состояния расширения аудиосигнала на основании алгоритма слепого разделения. ! 5. Устройство (100) по п.1, в котором блок (108) обнаружения выполнен с возможностью обнаружения состояния ...

МАНИПУЛИРОВАНИЕ ЗОНОЙ НАИЛУЧШЕГО ВОСПРИЯТИЯ ДЛЯ МНОГОКАНАЛЬНОГО СИГНАЛА

... 1. Устройство для модификации зоны наилучшего восприятия пространственного М-канального аудиосигнала, причем упомянутое устройство содержит: ! приемник (201) для приема N-канального аудиосигнала, N Подробнее

УСТРОЙСТВО И СПОСОБ ДЛЯ ПОВТОРНОГО ОТОБРАЖЕНИЯ ОТНОСЯЩИХСЯ К ЭКРАНУ ЗВУКОВЫХ ОБЪЕКТОВ

УСТРОЙСТВО ДЛЯ ОБРАБОТКИ СИГНАЛА, СПОСОБ ОБРАБОТКИ СИГНАЛА И ПРОГРАММА

АУДИОУСТРОЙСТВО И СПОСОБ ДЛЯ НЕГО

УСТРОЙСТВО ДЛЯ ФОРМИРОВАНИЯ ВЫХОДНОГО ПРОСТРАНСТВЕННОГО МНОГОКАНАЛЬНОГО АУДИОСИГНАЛА

... 1. Устройство (100) для определения пространственного выходного многоканального аудиосигнала, основанный на входном аудиосигнале, включающее декомпозитор (110) для разложения входного аудиосигнала, чтобы получить первую компоненту сигнала, имеющую первую семантическую особенность, и вторую компоненту сигнала, имеющую вторую семантическую особенность, отличающуюся от первой семантической особенности, где декомпозитор (110) выполнен с возможностью определения первой компоненты сигнала и/или второй компоненты сигнала на основе метода разделения перехода; блок рендеринга (120) для рендеринга первой компоненты сигнала, используя первую характеристику рендеринга, чтобы получить первый сигнал рендеринга, имеющий первую семантическую особенность, и для рендеринга второй компоненты сигнала, используя вторую характеристику рендеринга, чтобы получить второй сигнал рендеринга, имеющий вторую семантическую особенность, причем первая характеристика рендеринга и вторая характеристика рендеринга отличаются ...

УСТРОЙСТВО ОБРАБОТКИ ИНФОРМАЦИИ, СПОСОБ ОБРАБОТКИ ИНФОРМАЦИИ И ПРОГРАММА

Balancesteuerungsschaltung.

Verfahren zur Messung der Erhaltung stereophoner Audiosignale und Verfahren zur Erkennung gemeinsam codierter stereophoner Audiosignale

TESTING A PLURAL-CHANNEL AUDIO SIGNAL PROCESSING SYSTEM

Testing a plural-channel audio signal processing system

Spatially processing multichannel signals, processing module and virtual surround-sound system

Embodiments of a virtual surround-sound system and methods for simulating surround-sound are generally described herein. A processing module includes a spatial processor which spatially processes surround-left and surround-right channel signals and front-left and front-right channel signals and combines the spatially-processed signals for providing to drivers of centre speaker after crosstalk cancellation and combining with a centre-channel signal. In some embodiments, the processing module may include circuitry to cause the spatial processor to refrain from spatially processing either the front-left and front-right channel signals when front-left and/or front-right speakers are connected. The invention aims to reduce the set up complexity of prior art surround sound systems whilst making such systems less sensitive to the particular listening environment.

Simultaneous transmission of a plurality of audio data streams via a single communication link

Audio interface circuitry comprises a pulse length modulator, PLM 103, for transfer of multiple digital audio data streams over a single communications link, such as a single wire. The PLM is responsive to a plurality of data streams (PDM-R, PDM-L) of 1-bit audio data samples at a sample rate, to generate a stream of data pulses (MPDM) at the sample rate. The length of each said data pulse is dependent upon on a logical combination of the current audio data samples. The PLM may also be responsive to a first clock signal having a frequency equal to the sample rate, and a second clock signal having a frequency which is a multiple of the sample rate. The length of data pulses is preferably based on a selected number of cycles of the second clock signal. Circuitry for receiving and extracting the data is also disclosed. An interface 104 receives the stream of data pulses (MPDM) and data extraction circuitry 105, 106 determines the pulse length of said data pulse and determines a data value ...

Binaural synthesis

Audio system and method

Method for distributing a digital audio signal

A method for distributing a digital audio signal comprises transmitting timing information in a continuous channel (`the timing channel') that is synchronous to an audio clock (master clock) at a source. The timing channel includes information for both clock synchronization and sample synchronization. Audio sample data is transmitted in a separate channel (the data channel) that is asynchronous to the timing channel. The data channel may be optimized for data related parameters, such as bandwidth and robustness. The timing channel may be optimized for minimum clock jitter or errors in clock timing. The method is suitable for use in distributing stereo or surround sound signals to loudspeakers. A slave device receiving the timing channel may be equipped with a low bandwidth filter to filter out high frequency jitter, so that the jitter of the recovered slave clock is of the same order as the jitter in the master clock.

Method for distributing a digital audio signal

Control system

Spatial sound rendering

Microphone with advanced functionalities

Audio rendering with spatial metadata interpolation and source position information

A method of generating a spatial audio output by obtaining at least two audio signal sets, the positions of the audio signal sets, sound source position information, sound source energy values and a listener position; generating some form of residual background value based on the audio signal sets; and generating the spatial audio output for the listener based on the signals from the audio signal sets, the listener position, the sound source position, the sound source energy value and the residual background value. The audio signal sets may be microphones or arrays of microphones. The generated spatial audio output may enable the listener to have a six-degrees-of-freedom (6DoF) environment. The distance between the listener position and the microphone positions may be used to weight the audio signal sets. The sound source may be a prominent sound source which may be determined by having a sound energy greater than a threshold value.

SCHALTUNGSANORDNUNG ZUR GEGENSINNIGEN BEDAEMPFUNG ZWEIER TONKANAELE

PROCEDURE AND DEVICE FOR THE IMPROVEMENT OF THE AUDIO RECONSTRUCTION

ECONOMIC ONE LAUTHEITMESSUNG OF CODED AUDIO

SWITCHING CONFIGURATION FOR THE GEGENSINNIGEN DAMPING OF TWO CLAY/TONE CHANNELS

SWITCHING CONFIGURATION FOR THE GEGENSINNIGEN DAMPING OF TWO CLAY/TONE CHANNELS

Switching configuration for stereophonic sound tape-units for the automatic regulation of the reinforcement degree of the Aufsprechverstärker

Apparatus and method for providing individual sound zones

An apparatus for generating a plurality of loudspeaker signals from two or more audio source signals, wherein each of the two or more audio source signals shall be reproduced in one or more of two or more sound zones, and wherein at least one of the two or more audio source signals shall not be reproduced in at least one of the two more sound zones, wherein the apparatus comprises: an audio preprocessor configured to modify each of two or more initial audio signals to obtain two or more preprocessed audio signals, and a filter configured to generate the plurality of loudspeaker signals depending on the two or more preprocessed audio signals, wherein the audio preprocessor is configured to use the two or more audio source signals as the two or more initial audio signals, or wherein the audio preprocessor is configured to generate for each audio source signal of the two or more audio source signals an initial audio signal of the two more initial audio signals by modifying said audio source ...

System and method for adaptive audio signal generation, coding and rendering

Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object 5 based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of 10 reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent. The object position metadata contains the appropriate allocentric frame of reference information required to play the ...

Volume control device and distributed audio system

An apparatus and a method for converting a first parametric spatial audio signal into a second parametric spatial audio signal

An apparatus (300) for converting a first parametric spatial audio signal representing a first listening position or a first listening orientation in a spatial audio scene to a second parametric spatial audio signal (112, 114) representing a second listening position or a second listening orientation is described, the apparatus comprising: a spatial audio signal modification unit (301, 302) adapted to modify the first parametric spatial audio signal (212, 214) dependent on a change of the first listening position or the first listening orientation so as to obtain the second parametric spatial audio signal (212, 214), wherein the second listening position or the second listening orientation corresponds to the first listening position or the first listening orientation changed by the change.

Apparatus and method for audio content analysis, marking and summing

Method of interacting with the acoustical modal structure of a room

Method, signal processing unit, and computer program for mapping a plurality of input channels of an input channel configuration to output channels of an output channel configuration

A method for mapping a plurality of input channels of an input channel configuration to output channels of an output channel configuration comprises providing a set of rules associated with each input channel of the plurality of input channels, wherein the rules define different mappings between the associated input channel and a set of output channels. For each input channel of the plurality of input channels, a rule associated with the input channel is accessed, determination is made whether the set of output channels defined in the accessed rule is present in the output channel configuration, and the accessed rule is selected if the set of output channels defined in the accessed rule is present in the output channel configuration. The input channels are mapped to the output channels according to the selected rule.

Process for measuring the fidelity of stereophonic audio signals and process for recognising stereophonic audio signals coded together

SYSTEM AND TOOLS FOR ENHANCED 3D AUDIO AUTHORING AND RENDERING

Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

ECONOMICAL LOUDNESS MEASUREMENT OF CODED AUDIO

... ²²²Measuring the loudness of audio encoded in a bitstream that includes data from ²which an approximation of the power spectrum of the audio can be derived ²without fully decoding the audio is performed by deriving the approximation of ²the power spectrum of the audio from said bitstream without fully decoding the ²audio, and determining an approximate loudness of the audio in response to the ²approximation of the power spectrum of the audio. The data may include coarse ²representations of the audio and associated finer representations of the ²audio, the approximation of the power spectrum of the audio being derived from ²the coarse representations of the audio. In the case of subband encoded audio, ²the coarse representations of the audio may comprise scale factors and the ²associated finer representations of the audio may comprise sample data ²associated with each scale factor.² ...

APPARATUS AND METHOD FOR SCREEN RELATED AUDIO OBJECT REMAPPING

An apparatus for generating loudspeaker signals is provided. The apparatus comprises an object metadata processor (110) and an object renderer (120). The object renderer (120) is configured to receive an audio object. The object metadata processor (110) is configured to receive metadata, comprising an indication on whether the audio object is screen-related, and further comprising a first position of the audio object. The object metadata processor (110) is configured to calculate a second position of the audio object depending on the first position of the audio object and depending on a size of a screen, if the audio object is indicated in the metadata as being screen-related. The object renderer (120) is configured to generate the loudspeaker signals depending on the audio object and depending on position information. The object metadata processor (110) is configured to feed the first position of the audio object as the position information into the object renderer (120), if the audio ...

METHOD FOR PROCESSING AN AUDIO SIGNAL, SIGNAL PROCESSING UNIT, BINAURAL RENDERER, AUDIO ENCODER AND AUDIO DECODER

A method for processing an audio signal (504) in accordance with a room impulse response is described. The audio signal (504) is processed (502) with an early part of the room impulse response separate from a late reverberation of the room impulse response, wherein the processing (514) of the late reverberation comprises generating a scaled reverberated signal, the scaling (526) being dependent on the audio signal (504). The processed early part (506) of the audio signal (504) and the scaled reverberated signal are combined.

TRANSMITTING DEVICE, TRANSMITTING METHOD, RECEIVING DEVICE, AND RECEIVING METHOD

The purpose of the present invention is to enable good sound pressure adjustment of an object content on the reception side. An audio stream having coded data of a predetermined number of object contents is generated, and a predetermined format container including the audio stream is transmitted. Information indicating the allowable range of increase/decrease of sound pressure for each of the object contents is inserted into a layer of the audio stream and/or a layer of the container. On the reception side, processing for increasing/decreasing the sound pressure of each of the object contents within the allowable range is performed on the basis of the information.

METHOD FOR MEASURING THE CONSERVATION OF STEREOPHONIC AUDIO SIGNALS AND METHOD FOR IDENTIFYING JOINTLY CODED STEREOPHONIC AUDIO SIGNALS

Schaltungsanordnung zur Bestimmung der Richtung einer Schallquelle.

Kontrolleinrichtung für stereophonische Mehrkanalübertragungen

Device for inserting a single sound event in quadraphonic studio recording.

SIGNALING AUDIO RENDERING INFORMATION IN A BITSTREAM

Information processing apparatus, information processing method, and program

Spatial upmix controlled by renderer

Inspecting device for the transmission with multiple channels

METHOD OF SYNCHRONIZING AND PLAYBACK OF A MEDIA STREAM

Loudspeaker enclosure system control device for television display e.g. liquid crystal display, has reception and processing unit to process controls transmitted by remote selector of display to modify sound volume of system

L'invention concerne un dispositif de commande d'un système d'enceintes acoustiques (3, 4, 5) utilisé avec un écran de télévision (1), ledit écran étant du type équipé, en interne et/ou en externe, de moyens de restitution sonore spécifiques (9, 10), ledit écran étant associé à un boîtier de télécommande (2) émettant au moins un premier ensemble de commandes comprenant au moins une commande audio pour agir sur le volume sonore desdits moyens de restitution sonore spécifiques. Selon l'invention, le dispositif de commande comprend : des moyens (15) d'amplification ; des premiers moyens (12) de connexion, permettant de connecter les moyens d'amplification audit écran ; des seconds moyens (13) de connexion, permettant de connecter les moyens d'amplification auxdites enceintes acoustiques ; des moyens (16) réception et de traitement dudit premier ensemble de commandes, agissant sur lesdits moyens d'amplification de façon à modifier le volume sonore dudit système d'enceintes acoustiques en fonction ...

DEVICE TO MAKE CORRESPOND STATIC CHARACTERISTICS OF VOLUME IN D.C. CURRENT OF TWO AUDIO CHANNELS

Electroacoustic instrument transposing device - has photoresistive switching elements to extend change over and control progression between notes

Recording studio channel selection/mixer having parallel audio channel inputs multiple command selectable switching bank and amplifier/headset output.

Appareil répartiteur et mixeur d'écoute à plusieurs voies, comportant plusieurs entrées (4) de signaux électriques audio provenant de diverses sources sonores et plusieurs sorties (5) pouvant être connectées respectivement à des casques à écouteurs (6) portés par des auditeurs, remarquable en ce que chaque voie d'écoute (A... N) comporte un étage de commutation comprenant plusieurs interrupteurs (7) branchés en parallèle et reliés respectivement aux différentes entrées (4), un commutateur (9) relié aux différents interrupteurs (7) et un moyen (11) de commande du commutateur (9), permettant à chaque auditeur de sélectionner à volonté l'entrée des signaux audio qu'il désire écouter, et un étage d'amplification (8) relié en parallèle aux sorties des interrupteurs (7) et dont la sortie est connectée à la sortie (5) de l'appareil relié au casque (6).

이형부, 인공 두부 및 이들을 이용한 측정 시스템 및 측정 방법

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

System and method for adaptive audio signal generation, coding and rendering

Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent. The object position metadata contains the appropriate allocentric frame of reference information required to play the sound ...

LAYERED CODING FOR COMPRESSED SOUND OR SOUND FIELD REPRESENTATIONS

LAYERED CODING FOR COMPRESSED SOUND OR SOUND FIELD REPRESENTATIONS The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation. The method comprises sub-dividing the plurality of components into a plurality of groups of components and assigning each of the plurality of groups to a respective one of a plurality of hierarchical layers, the number of groups corresponding to the number of layers, and the plurality of layers including a baselayer and one or more hierarchical enhancement layers, adding the basic side information to the base layer ...

Economical loudness measurement of coded audio

DETERMINATION OF SYSTEM CHARACTERISTICS

An apparatus is disclosed for determining characteristics of a linear system that is either time invariant or varying in time. The apparatus comprises: a probe signal splitter operable to receive a probe signal for probing a system under test, and to split the probe signal into first and second probe signal components such that the second probe signal component is a phase- shifted version of the first probe signal component with a phase difference θ satisfying θ MOD 180° < > 0°; a response signal splitter operable to receive a response signal from the system under test comprising a response of the system to the probe signal, and to split the response signal into first and second response signal components such that the second response signal component is a phase-shifted version of the first response signal component with a phase difference corresponding to the phase difference of the first and second probe signal components; a crosslation calculator operable to perform: a first crosslation ...

PROGRAM USED FOR TERMINAL APPARATUS, SOUND APPARATUS, SOUND SYSTEM, AND METHOD USED FOR SOUND APPARATUS

A sound apparatus comprises: an acceptance unit that accepts the input of input audio signals from the exterior; a communication unit that receives, from a terminal apparatus, first direction information indicating a first direction that is a direction in which a virtual sound source is located; a position information generation unit that generates, on the basis of viewing/listening position information indicating a viewing/listening position, of the first direction information and of boundary information indicating the boundary of a space where the virtual sound source is located, virtual sound source position information indicating the position of the virtual sound source on the boundary; a signal generation unit that imparts, on the basis of speaker position information indicating the positions of a plurality of speakers, of the viewing/listening position information and of the virtual sound source position information, acoustic effects to the input audio signals such that sounds will ...

VOLUME CONTROL IN MOVIE THEATERS

A motion picture soundtrack reproduction system has a center front soundtrack channel and a plurality of other soundtrack channels. A volume control adjusts the gain of all the channels. The volume control has a range of settings from a minimum to a maximum, the gain of the center front channel having substantially a first relationship to the volume control settings and the gain of the other channels having substantially a second relationship to the volume control settings, the relationships being such that for a range of volume control settings less than a first setting the gain of the center front channel remains substantially constant while the gain of the other channels decreases as the setting decreases or decreases more gradually than the gain of the other channels as the setting decreases.

DATA-DRIVEN SOFTWARE ARCHITECTURE FOR DIGITAL SOUND PROCESSING AND EQUALIZATION

A digital sound processing design system for a vehicle audio system includes a computer and a design tool that is run by the computer. The design tool allows a user to define sound processing criteria that is stored in a template file. An audio signal processor is connected to the first and second real channel inputs of an audio source. Memory that is coupled to the audio signal processor stores the template file. The sound processing engine that is coupled to the audio signal processor and the memory reads the template file at run-time to obtain the sound processing criteria. The sound processing engine applies the sound processing criteria to the first and second real channel inputs. The design tool allows a user to create virtual channel inputs and outputs that are based, in part, on the first and second real channel inputs.

AUDIO ACCESSIBILITY

An audio delivery method. An image of a listening area is captured and processed to locate a position of a listener in the room. A stored listener profile associated with the listener is retrieved and audio characteristics are established based on the listener's profile. A directional beam of audio is directed toward the listener's ears and the directional beam is adjusted to track movement of the listener. This abstract is not to be considered limiting, since other embodiments may deviate from the features described in this abstract.

Passive and Active Virtual Height Filter Systems for Upward Firing Drivers

Embodiments are directed to a virtual height filter for use with or in an upward-firing speaker system that reflects sound off a ceiling to a listening location at a distance from a speaker, and that provides height cues to reproduce audio objects that have overhead audio components. A virtual height filter based on a directional hearing model is applied to the upward-firing driver signal to improve the perception of height for audio signals transmitted by the virtual height speaker to provide optimum reproduction of the overhead reflected sound. The virtual height filter is provided by any one or combination of analog or digital filter circuits, or mechanical structures including speaker grill, enclosure, or driver design or configuration.

Method and apparatus for automatically setting speaker mode in audio/video system

A method and apparatus for automatically determining a characteristic of a speaker and automatically setting a speaker mode in an audio/video system. The method includes: detecting a current for operating the speaker by inputting a predetermined signal; measuring an impedance characteristic of the speaker in accordance with a frequency change based on the detected current; determining a speaker type based on the measured impedance characteristic; and setting a speaker mode based on an impedance characteristic curve of the discriminated speaker type.

Auto voice changing apparatus

An auto voice changing apparatus receives music from a stereo or multi-sound, digital or analog recording and either plays the recorded music as is, or uses the recorded music as instrumental accompaniment by eliminating the vocal sounds from the reproduced music and substituting vocal sounds provided through a microphone. The control is based solely on a single auto voice switch, control information from the recording, and detection of vocal sounds from the microphone.

Audio reproducing apparatus

In an audio reproducing apparatus, first and second filters convolute impulse responses corresponding to transfer functions from a position where a right-hand sound source is located to the right and left ears of the listener into an audio signal, respectively, and third and fourth filters convolute impulse responses corresponding to transfer functions from a position where a left-hand sound source is located to the right and left ears of the listener into another audio signal, respectively. Fifth and sixth filters extract low-frequency components of the audio signal, and seventh and eighth filters extract low-frequency components of the another audio signal. The output signals of the first, third, fifth, and seventh filters are added, and the output signals of the second, fourth, the sixth, and eighth filters are added.

Binaural synthesis

Embodiments relate to obtaining filter coefficients for a binaural synthesis filter; and applying a compensation filter to reduce artifacts resulting from the binaural synthesis filter; wherein the filter coefficients and compensation filter are configured to be used to obtain binaural audio output from a monaural audio input. The filter coefficients and compensation filter may be applied to the monaural audio input to obtain the binaural audio output. The compensation filter may comprise a timbre compensation filter.

AUTOMATED USER/SENSOR LOCATION RECOGNITION TO CUSTOMIZE AUDIO PERFORMANCE IN A DISTRIBUTED MULTI-SENSOR ENVIRONMENT

A wireless device is described. The wireless device includes at least two microphones on the wireless device. The microphones are configured to capture sound from a target user. The wireless device also includes processing circuitry. The processing circuitry is coupled to the microphones. The processing circuitry is configured to locate the target user. The wireless device further includes a communication interface. The communication interface is coupled to the processing circuitry. The communication interface is configured to receive external device microphone audio from at least one external device microphone to assist the processing circuitry in the wireless device to locate the target user.

Binaural rendering method and apparatus for decoding multi channel audio

Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal.

Audio processor

An audio information detector extracts frequency information on audio data from a packet called an ASP in the HDMI standard and outputs the extracted frequency information to a frequency divider as audio information. The frequency divider determines a frequency division ratio based on the audio information, divides the frequency of a PLL clock signal output from an analog PLL circuit by the frequency division ratio and outputs the resultant signal as a comparison clock signal. The analog PLL circuit performs feedback control such that the comparison clock signal and a reference clock signal are synchronized with each other, and generates an audio clock signal obtained by performing frequency multiplication or division on the reference clock signal.

Processing Audio Signals

A method, apparatus and computer program including obtaining a first audio signal from at least a first microphone located at a first distance from an audio source; obtaining a local audio signal from a local microphone wherein the local microphone is located closer to the audio source than the first microphone; identifying the audio source using the first audio signal and the local audio signal; using the identified audio source to determine a delay between the first audio signal and the local audio signal: and using the determined delay to determine the length of the first audio signal and the local audio signal to be made available for processing so as to enable temporal alignment of at least the first audio signal and the local audio signal.

AUDIO SYSTEM AND METHOD

Embodiments relate to, for a scene comprising a representation of at least one object and at least one sound source: obtaining a decomposition of the at least one object, the decomposition comprising at least one geometric component; modelling at least one interaction of the at least one object and the at least one sound source using the at least one geometric component; and, in dependence on the modelling of the at least one interaction, processing an audio input associated with the at least one sound source to obtain an audio output.

Apparatus and method for generating a plurality of parametric audio streams and apparatus and method for generating a plurality of loudspeaker signals

An apparatus for generating a plurality of parametric audio streams from an input spatial audio signal obtained from a recording in a recording space has a segmentor and a generator. The segmentor is configured for providing at least two input segmental audio signals from the input spatial audio signal, wherein the at least two input segmental audio signals are associated with corresponding segments of the recording space. The generator is configured for generating a parametric audio stream for each of the at least two input segmental audio signals to obtain the plurality of parametric audio streams.

Speaker normalization system

System, method, and various embodiments for providing a speaker normalization system are described herein. An embodiment operates by identifying a plurality of audio output devices connected to an audio source. For each audio output device, a set of technical capabilities and a relative position of the audio output device is identified. An audio configuration is calculated based on both the set of technical capabilities and the relative position of each audio output device, and which audio output devices are identified. The calculated audio configuration is provided to each identified audio output device. Audio is provided to each of the identified audio output devices.

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

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

Меthоd аnd аppаrаtus fоr аutоmаtiсаllу sеtting spеаkеr mоdе in аudiо/vidео sуstеm

А mеthоd аnd аppаrаtus fоr аutоmаtiсаllу dеtеrmining а сhаrасtеristiс оf а spеаkеr аnd аutоmаtiсаllу sеtting а spеаkеr mоdе in аn аudiо/vidео sуstеm. Тhе mеthоd inсludеs: dеtесting а сurrеnt fоr оpеrаting thе spеаkеr bу inputting а prеdеtеrminеd signаl; mеаsuring аn impеdаnсе сhаrасtеristiс оf thе spеаkеr in ассоrdаnсе with а frеquеnсу сhаngе bаsеd оn thе dеtесtеd сurrеnt; dеtеrmining а spеаkеr tуpе bаsеd оn thе mеаsurеd impеdаnсе сhаrасtеristiс; аnd sеtting а spеаkеr mоdе bаsеd оn аn impеdаnсе сhаrасtеristiс сurvе оf thе disсriminаtеd spеаkеr tуpе.

Меthоd аnd аppаrаtus fоr аutоmаtiсаllу sеtting spеаkеr mоdе in аudiо/vidео sуstеm

А mеthоd аnd аppаrаtus fоr аutоmаtiсаllу dеtеrmining а сhаrасtеristiс оf а spеаkеr аnd аutоmаtiсаllу sеtting а spеаkеr mоdе in аn аudiо/vidео sуstеm. Тhе mеthоd inсludеs: dеtесting а сurrеnt fоr оpеrаting thе spеаkеr bу inputting а prеdеtеrminеd signаl; mеаsuring аn impеdаnсе сhаrасtеristiс оf thе spеаkеr in ассоrdаnсе with а frеquеnсу сhаngе bаsеd оn thе dеtесtеd сurrеnt; dеtеrmining а spеаkеr tуpе bаsеd оn thе mеаsurеd impеdаnсе сhаrасtеristiс; аnd sеtting а spеаkеr mоdе bаsеd оn аn impеdаnсе сhаrасtеristiс сurvе оf thе disсriminаtеd spеаkеr tуpе.

Sound signal processing apparatus and sound signal processing method

A satisfactory sound volume sensation from a hearing point of view can be obtained, and an effective sound volume correction operation can be obtained with an algorithm as simple as possible. Correction step 0 (correction function off, level correction value 0) to correction step 16 (level correction value - 16 dB) are defined. When a first condition that is defined in such a manner as to correspond to a state in which distortion occurs is satisfied, the correction step is made to proceed by one step from the current correction step (-1 dB is added to a level correction value). When a second condition that is defined in such a manner as to correspond to a state in which distortion does not occur is satisfied, the correction step is made to return by one step from the current correction step (-1 dB is subtracted from the level correction value).

BALANCE-CORRECTING CIRCUIT OF AUDIO SIGNAL

PURPOSE: To automatically correct the balance of L and R signals and to execute optimum audio listening by detecting the average values of audio L and R channels, comparing them and impressing inverse proportion outputs. CONSTITUTION: The average value of the L signal and the R signal of the L channel 10 and the R channel 12 are detected by respective detectors 14 and 16. That is, they are the RMS values of the L and R signals which are detected by the detectors 14 and 16 so as to be a DC level. Therefore, the outputs of the detectors 14 and 16 are inputted to a differential comparison part 25. A first comparing signal of the comparing part 25 is output-generated by the collector of TR2, a second comparing signal is generated by the collector of TR1, they are inputted to respective LPFs 32 and 34, and only the power which is equal to below an interrupting frequency is made to pass through. The outputs of LPFs 32 and 34 are respectively inputted to gain control parts 36 and 38, the L signal ...

СПОСОБ И СИСТЕМА ДЛЯ ВЫРАВНИВАНИЯ ГРОМКОГОВОРИТЕЛЯ В КОМНАТЕ

Изобретение относится к области аудио- и звуковоспроизведения, в частности к способам и системам для выравнивания частотных характеристик громкоговорителя в комнате с целью его адаптации. Техническим результатом является улучшение качества воспринимаемого звуковоспроизведения громкоговорителя в положении для прослушивания. Указанный технический результат достигается тем, что способ выравнивания частотной характеристики громкоговорителя, расположенного в комнате, включает стадию измерения передаточной функции в положении для прослушивания (L) от электрического входа первого громкоговорителя (L1) до звукового давления в положении для прослушивания (LP) в комнате, стадию определения общей передаточной функции (G), представляющей пространственное среднее уровня звукового давления в комнате, создаваемого первым громкоговорителем (L1), стадию определения верхнего предела усиления (UGL) как функции частоты, причем верхний предел усиления (UGL) основывают на инверсии общей передаточной функции ...

КОНЦЕПЦИЯ ГЕНЕРИРОВАНИЯ СИГНАЛА ПОНИЖАЮЩЕГО МИКШИРОВАНИЯ

Изобретение относится к области обработки звуковых сигналов. Технический результат – повышение вычислительной эффективности понижающего микширования входных сигналов за счет предотвращения создания искажений. Устройство обработки звуковых сигналов для понижающего микширования первого входного сигнала () и второго входного сигнала () в сигнал () понижающего микширования, в котором первый входной сигнал () и второй входной сигнал (), по меньшей мере, частично коррелированы, содержит: блок выделения несходства, выполненный с возможностью приема первого входного сигнала () и второго входного сигнала (), а также вывода выделенного сигнала (), который менее коррелирован по отношению к первому входному сигналу (), чем второй входной сигнал (), и блок объединения, выполненный с возможностью объединения первого входного сигнала () и выделенного сигнала () с целью получения сигнала () понижающего микширования, причем блок выделения несходства содержит блок оценки сходства и блок уменьшения сходства ...

СИСТЕМА, УСТРОЙСТВО И СПОСОБ ДЛЯ СОВМЕСТИМОГО ВОСПРОИЗВЕДЕНИЯ АКУСТИЧЕСКОЙ СЦЕНЫ НА ОСНОВЕ АДАПТИВНЫХ ФУНКЦИЙ

Изобретение относится к средствам для воспроизведения акустической сцены. Технический результат заключается в повышении эффективности обработки аудиосигнала. Принимают сигнал прямых компонент, содержащий компоненты прямых сигналов из двух или более исходных аудиосигналов. Принимают сигнал диффузных компонент, содержащий компоненты диффузных сигналов из упомянутых двух или более исходных аудиосигналов. Принимают информацию направления, при этом упомянутая информация направления зависит от направления прибытия компонент прямых сигналов из упомянутых двух или более исходных аудиосигналов. Генерируют один или более обработанных диффузных сигналов в зависимости от сигнала диффузных компонент. Для каждого выходного аудиосигнала из упомянутых одного или более выходных аудиосигналов определяют в зависимости от направления прибытия усиление прямого звука, применяют упомянутое усиление прямого звука к сигналу прямых компонент, чтобы получить обработанный прямой сигнал. 3 н. и 14 з.п. ф-лы, 21 ил.

ВОСПРОИЗВЕДЕНИЕ НА НИЖНЕМ УРОВНЕ

Изобретение относится к средствам для воспроизведения многоканальных аудиоданных. Технический результат заключается в повышении эффективности воспроизведения многоканальных аудиоданных. Процессор нижнего уровня обрабатывает по меньшей мере один канал или более каналов относящихся к объектам аудиоданных или многоканальных аудиоданных, принадлежащих нижнему уровню. Обычный уровень является уровнем прослушивания, находящимся на высоте головы слушателя с точки зрения слушателя, причем обычный уровень находится выше нижнего уровня с точки зрения слушателя. Процессор LFE/басов подает низкочастотные участки одного или более каналов обычного уровня на сабвуфер через выход сабвуфера. Процессор нижнего уровня выполнен с возможностью подачи по меньшей мере одного сигнала, принадлежащего одному каналу или более каналам нижнего уровня, на упомянутый выход сабвуфера для воспроизведения одного или более каналов нижнего уровня с использованием упомянутого сабвуфера. Частоты частотного участка одного или ...

СПОСОБ И УСТРОЙСТВО ДЛЯ ОПРЕДЕЛЕНИЯ ПОЗИЦИИ МИКРОФОНА

Изобретение относится к средствам для определения позиции микрофона. Технический результат заключается в повышении точности определения позиции микрофона. Устройство содержит приемник (203), принимающий многоканальный сигнал, содержащий два канала для воспроизведения первым динамиком (101) в первой позиции и вторым динамиком (103) во второй позиции соответственно. Первый генератор (207) сигналов генерирует коррелированный сигнал и второй генератор (209) сигналов генерирует некоррелированный сигнал из многоканального сигнала, причем сигналы содержат соответственно коррелированные и некоррелированные компоненты сигнала для каналов. Приемник (201) принимает сигнал микрофона от микрофона (107). Первое средство (213) корреляции определяет первый корреляционный сигнал из корреляции сигнала микрофона и коррелированного сигнала, и второе средство (215) корреляции определяет второй корреляционный сигнал из корреляции сигнала микрофона и некоррелированного сигнала. Средство (219) оценки позиции оценивает ...

УСТРОЙСТВО И СПОСОБ ПРЕОБРАЗОВАНИЯ ПЕРВОГО И ВТОРОГО ВХОДНЫХ КАНАЛОВ, ПО МЕНЬШЕЙ МЕРЕ, В ОДИН ВЫХОДНОЙ КАНАЛ

Изобретение относится к области кодирования аудиосигналов и предназначено для преобразования первого и второго входных каналов в один выходной канал, и в частности, предназначено для использования при преобразовании формата между различными конфигурациями каналов громкоговорителей. Технический результат - улучшенное воспроизведение звука в случае преобразования формата между различными конфигурациями каналов громкоговорителей. Пространственное кодирование аудио начинается с множества исходных входных, например пяти или семи входных каналов, которые идентифицируются посредством их размещения в компоновке для воспроизведения в качестве левого канала, центрального канала, правого канала, левого канала объемного звучания, правого канала объемного звучания и канала улучшения низких частот (LFE). В устройстве каждый входной канал и каждый выходной канал имеет направление, в котором расположен ассоциированный громкоговоритель относительно центральной позиции слушателя, при этом устройство выполнено ...

Apparatus for determining a spatial output multi-channel audio signal

An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.

Integrated speaker and home theater sound system having the same

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

Acoustic Localization of a Speaker

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

Parametric joint-coding of audio sources

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Method and apparatus for reproducing three-dimensional sound

Stereophonic sound is reproduced by acquiring image depth information indicating a distance between at least one object in an image signal and a reference location, acquiring sound depth information indicating a distance between at least one sound object in a sound signal and a reference location based on the image depth information, and providing sound perspective to the at least one sound object based on the sound depth information.

Modular audio systems and related assemblies and methods

Modular audio systems comprise two speaker assemblies and a wiring system. Each speaker assembly may comprise a speaker and an audio jack integral to each speaker assembly. The wiring system comprises a first wiring assembly comprising two audio jacks configured to connect to the audio jacks of the two speaker assemblies and two wires connected to the two audio jacks at first ends of the two wires. Headphone assemblies may comprise two speaker assemblies and a headband configured for removable attachment to the speaker assemblies. Each speaker assembly may comprise an attachment structure configured for attachment to another device or structure, wherein the attachment structure of each speaker assembly of the two speaker assemblies comprises a frustoconical surface and two first attachment features comprising elongated features on the frustoconical surface.

Binaural audio signal-based applications

Applications for detecting binaural sounds and customizing binaural sounds for users. A system for detecting binaural sounds according to an embodiment includes: an input system for obtaining a left side audio signal and a right side audio signal within an environment; a binaural sound detector for evaluating the left side audio signal and the right side audio signal to determine whether a binaural sound is present; and an alarm system for generating an alert when the binaural sound is detected.

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

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

METHOD AND SYSTEM FOR MULTI-CHANNEL MIXING FOR TRANSMISSION OF AUDIO OVER A NETWORK

A method and system for providing remote access to multi-channel audio by periodically polling channels of audio produced by, e.g., an application program by calling an Application Programming Interface (API). A method performs the polling to retrieve audio data from multiple channels and to mix the multiple channels into a mixed multichannel audio that is communicated to a remote computing device. The method transmits a sample minimum duration of audio data retrieved from all channels during at polling interval to provide low latency transmission of audio to remotely connected computing devices. 1. A method of communicating mixed multichannel audio associated with a source to a remote client computing device as single channel audio using a remote access server , comprising:determining a source-depleted time for each channel of the multichannel audio;determining if there is audio data present on all channels at the source;determining a sample minimum amount of audio that is present for all of the channels;mixing the sample minimum amount of audio for each of the channels of the multichannel audio into the single channel audio; andcommunicating the single channel audio to the remote client computing device.2. The method of claim 1 , wherein the determining if there is audio data present at the source is performed at a predetermined interval.3. The method of claim 2 , wherein the predetermined interval is 15 ms.4. The method of claim 2 , further comprising updating the source-depleted time for each channel at each predetermined interval.5. The method of claim 1 , wherein if all channels do not have data claim 1 , the method further comprising:determining if the source-depleted time has elapsed for any of the channels; andif not, waiting for a next call interval.6. The method of claim 5 , wherein if the source-depleted time has elapsed on a channel claim 5 , then:determining if the remaining channels have remaining audio data; andmixing the sample minimum duration of ...

Signal processing device, sound-reproduction system, and sound reproduction method

A signal processing device includes: a processor; and a memory having instructions. The instructions, when executed by the processor, cause the signal processing device to perform operations. The operations include performing a modulation processing of modulating a sound signal by using a modulation parameter based on an interaural phase difference at a listening position of the sound signal.

Earphone and sound channel control method thereof

An earphone is provided. The earphone includes a first speaker, an audio processor, a sensor, and a controller. The audio processor provides a left sound channel signal and a right sound channel signal according to a stereo signal. The sensor senses a movement path of the earphone. The controller selectively provides one of the left sound channel signal and the right sound channel signal to the first speaker according to the movement path.

Multi-Channel Signal Encoding Method, Multi-Channel Signal Decoding Method, Encoder, and Decoder

A multi-channel signal decoding method includes obtaining a downmixed signal of a first channel signal and a second channel signal in a multi-channel signal, obtaining parameter indication information, obtaining a reverberation gain parameter corresponding to each of the part of subbands according to the parameter indication information; and obtaining a reconstructed first channel signal and a reconstructed second channel signal according to the downmixed signal and the reverberation gain parameter.

MICROPHONE AND MANUFACTURING METHOD THEREOF

A microphone includes: a case that is vibrated by a vibration signal, a sound inlet through which a sound signal is input being formed at a portion of the case; a first sound element that is formed in the case at a position corresponding to the sound inlet and receives the sound signal and the vibration signal to output a first initial signal; a second sound element that is formed to be adjacent to the first sound element and receives the vibration signal to output a second initial signal; and a semiconductor chip that is connected to the first sound element and the second sound element and receives the first initial signal and the second initial signal to output a final signal. 1. A microphone comprising:a case that is vibrated by a vibration signal, a sound inlet through which a sound signal is input being formed at a portion of the case;a first sound element that is formed in the case at a position corresponding to the sound inlet and receives the sound signal and the vibration signal to output a first initial signal;a second sound element that is formed to be adjacent to the first sound element and receives the vibration signal to output a second initial signal; anda semiconductor chip that is connected to the first sound element and the second sound element and receives the first initial signal and the second initial signal to output a final signal.2. The microphone of claim 1 , wherein the semiconductor chip: i) divides the first initial signal into a sound signal and a vibration signal claim 1 , ii) modulates a phase of the second initial signal claim 1 , iii) merges the first initial signal with the divided sound signal and vibration signal claim 1 , and iv) merges the second initial signal with the phase-modulated signal to cancel the vibration signal and extract the sound signal.3. The microphone of claim 1 , wherein an air passage is formed at a side of a lower portion of the second sound element.4. The microphone of claim 1 , wherein the case includes:a ...

METHOD FOR CAPTURING AND PLAYBACK OF SOUND ORIGINATING FROM A PLURALITY OF SOUND SOURCES

The invention discloses a method for capturing and for play-back of sound originating from a plurality of sources. It also includes a computer program product having an audio file adapted to receive and play back such sound. Basically, sound originating from each sound source is recorded on individual tracks. To preserve the spatial distribution and the movement of the sound sources, the current positions of the sound sources are also recorded relative to at least one listening position. Furthermore, movements of one or more listeners during playback can be tracked and used for rendering the spatial acoustic field during playback tailored to the current position of the listener(s). 114-. (canceled)15. A method for playback of recorded sound associated with a plurality of sound sources , the method comprising: 'a number of recording tracks, each recording track having recorded sound originated from one of the sound sources, and repeatedly stored positions associated with the sound sources, each stored position representing a current position of one of the sound sources relative to at least one listening position;', 'providing an audio file, wherein the audio file comprisesproviding an audio playback system including a number of playback channels, wherein the playback system includes a computing unit programmed to generate a spatial acoustic field based on the recorded sounds and repeatedly stored positions included in the audio file; andplayback of the spatial acoustic field on the audio playback system.16. The method according to claim 15 , wherein generating the spatial acoustic field is adapted to the number of the playback channels during playback.17. The method according to claim 15 , further comprising providing a sensor adapted to track a current position of at least one listener.18. The method according to claim 17 , wherein generating the spatial acoustic field includes adapting the repeatedly stored positions to the tracked current position of the at least ...

Computer program and method of determining a personalized head-related transfer function and interaural time difference function

A method of estimating an individualized head-related transfer function (HRTF) and an individualized interaural time difference function (ITDF) of a particular person, comprising the steps of: a) obtaining a plurality of data sets (Li, Ri, Oi), each comprising a left and a right audio sample (Li, Ri) from a pair of in-ear microphones, and orientation information (Oi) from an orientation unit, measured in a test-arrangement whereby an acoustic test signal is rendered via a loudspeaker; b) storing the data sets in a memory; c) estimating the directions of the loudspeaker relative to the person based on the orientation data and the audio data; d) estimating the ITDF based on the data sets and on the estimated relative position/orientation; e) estimating the HRTF, based on the data sets and based on the estimated relative position/orientation.

Sound Masking System

A sound masking system according to the invention is disclosed in which one or more sound masking loudspeaker assemblies are coupled to one or more electronic sound masking signal generators. The loudspeaker assemblies in the system of the invention have a low directivity index and preferably emit an acoustic sound masking signal that has a sound masking spectrum specifically designed to provide superior sound masking in an open plan office. Each of the plurality of loudspeaker assemblies is oriented to provide the acoustic sound masking signal in a direct path into the predetermined area in which masking sound is needed. In addition, the sound masking system of the invention can include a remote control function by which a user can select from a plurality of stored sets of information for providing from a recipient loudspeaker assembly an acoustic sound masking signal having a-selected sound masking spectrum. 1a plurality of loudspeaker assemblies, each loudspeaker assembly coupled to one or more sources of an electrical sound masking signal,wherein each of the plurality of loudspeaker assemblies has a voice coil coupled to an audio emitter operative to emit an acoustic sound masking signal corresponding to said electrical sound masking signal, wherein each said audio emitter is a cone emitter, wherein each of the plurality of loudspeaker assemblies has a low directivity index, and wherein each of the plurality of loudspeaker assemblies is constructed and oriented to provide the acoustic sound masking signal to produce direct field sound masking in said predetermined area; andwherein, in said plurality of loudspeaker assemblies each having a low directivity index, each of said audio emitters has an aperture and each of the plurality of loudspeaker assemblies comprises a reflector positioned relative to said audio emitter aperture so as to scatter high frequency sounds in said acoustic sound masking signal.. A direct field sound masking system for providing direct ...

Display control apparatus, display control method, and program

A display control apparatus including a display control portion configured to display, on a display portion, a position of an array speaker, a time waveform that corresponds to at least one sound source, and a trajectory indicating a variation in a localization point of the sound source in a predetermined space.

BIOMETRIC, PHYSIOLOGICAL OR ENVIRONMENTAL MONITORING USING A CLOSED CHAMBER

At least one embodiment is directed to an earphone that includes biometric sensors, where the biometric sensor can detect levels of gas concentration. Where the gas levels are reported to a user if exceeding a threshold value.

Apparatus and Method for Providing Individual Sound Zones

An apparatus for generating a plurality of loudspeaker signals from two or more audio source signals is provided. Each of the two or more audio source signals shall be reproduced in one or more of two or more sound zones, and at least one of the two or more audio source signals shall not be reproduced in at least one of the two more sound zones. An audio preprocessor is configured to modify each initial audio signal of the two or more initial audio signals depending on a signal power or a loudness of another initial audio signal of the two or more initial audio signals. A filter is configured to generate the plurality of loudspeaker signals depending on in which of the two or more sound zones the two or more audio source signals shall be reproduced and depending on in which of the two or more sound zones the two or more audio source signals shall not be reproduced.

HEARING INSTRUMENT AND METHOD OF PROVIDING SUCH HEARING INSTRUMENT

A method of providing a hearing instrument includes: providing at a manufacturer the hearing instrument with a firmware which comprises a plurality of selectable functional properties; distributing the hearing instrument from the manufacturer to a middleman, wherein the hearing instrument is configured to be programmed by a programming station at the middleman, wherein the programming station is configured for selecting at least one of the plurality of selectable functional properties according to an input from the middleman, wherein the programming station is also configured to program the firmware of the hearing instrument according to the selecting to provide a programmed hearing instrument; billing an end user for the programmed hearing instrument, wherein the billing reflects the selecting; and receiving, at the manufacturer, data regarding the selecting from the middleman. 1. A method of providing a hearing instrument , comprising:providing at a manufacturer the hearing instrument with a firmware which comprises a plurality of selectable functional properties;distributing the hearing instrument from the manufacturer to a middleman, wherein the hearing instrument is configured to be programmed by a programming station at the middleman, wherein the programming station is configured for selecting at least one of the plurality of selectable functional properties according to an input from the middleman, wherein the programming station is also configured to program the firmware of the hearing instrument according to the selecting to provide a programmed hearing instrument;billing an end user for the programmed hearing instrument, wherein the billing reflects the selecting; andreceiving, at the manufacturer, data regarding the selecting from the middleman.2. The method according to claim 1 , wherein the firmware of the hearing instrument comprises substantially all functional properties available to the manufacturer.3. The method according to claim 1 , wherein at ...

Merging Audio Signals with Spatial Metadata

Apparatus for mixing at least two audio signals, at least one audio signal associated with at least one parameter, and at least one second audio signal further associated with at least one second parameter, wherein the at least one audio signal and the at least one second audio signal are associated with a sound scene and wherein the at least one audio signal represent spatial audio capture microphone channels and the at least one second audio signal represents an external audio channel separate from the spatial audio capture microphone channels, the apparatus comprising: a processor configured to generate a combined parameter output based on the at least one second parameter and the at least one parameter; and a mixer configured to generate a combined audio signal with a same number or fewer number of channels as the at least one audio signal based on the at least one audio signal and the at least one second audio signal, wherein the combined audio signal is associated with the combined parameter. 128-. (canceled)29. An apparatus configured to mix at least one first audio signal , accompanied with associated at least one first parameter , and at least one second audio signal , associated with at least one second parameter , where the apparatus comprises a processor configured to:generate a combined audio signal based, at least partially, upon the at least one first audio signal and the at least one second audio signal, where the combined audio signal comprises a fewer number of channels than a combined number of channels of the at least one first audio signal and the at least one second audio signal; andgenerate a combined parameter, where the combined parameter is based, at least partially, on the at least one first parameter and the at least one second parameter, where the combined parameter comprises one or more first elements based on the at least one first parameter and comprises one or more second elements based on the at least one second parameter,where the ...

SMART HUB

An intelligent hub for interfacing with other devices and performing smart audio or video source selection. 114.-. (canceled)15. A wireless speaker device for executing commands based on voice recognition and detecting the presence of other wireless speaker devices , the wireless speaker device comprising:a Wi-Fi radio;a Bluetooth radio;one or more microphones;at least one loudspeaker;a signal generator for providing one or more audio information signals;an audio information interface for providing the one or more audio information signals for use in the generation of respective acoustic signals to respective loudspeakers; and detect the presence of a second wireless speaker device, utilizing the Bluetooth radio;', 'communicate with the second wireless speaker device utilizing the Wi-Fi radio and/or the Bluetooth radio, to provide simultaneous audio playback of streaming content utilizing both the at least one loudspeaker located within the wireless speaker device and at least one loudspeaker local to the second wireless speaker device;', 'communicate with one or more remote servers using the Wi-Fi radio to receive streaming content for playback from a plurality of content sources;', 'receive acoustic signals using the one or more microphones;', 'perform voice command recognition on the acoustic signals, to determine a connected speaker command; and', 'perform one of a plurality of functions in response to the at least one connected speaker command, wherein the plurality of functions includes selecting one of the plurality of content sources for playback to both of the wireless speaker devices at the same time., 'one or more processors configured to execute program code to16. The device of wherein the acoustic signals include the streaming content.17. The device of wherein the acoustic signals comprise sounding signals that are inaudible signals embedded in audio signals broadcast by at least one of the respective speaker devices claim 15 , and the processor is ...

HEADSET WITH USER CONFIGURABLE NOISE CANCELLATION vs AMBIENT NOISE PICKUP

A system comprises automatic noise cancellation circuitry and interface circuitry operable to provide an interface via which a user can configure which sounds said automatic noise cancelling circuitry attempts to cancel and which sounds said automatic noise cancelling circuitry does not attempt to cancel. The interface circuitry may be operable to provide an interface via which a user can select a sound to whitelist or blacklist. The interface circuitry may be operable to provide an interface via which a user can increase or decrease an amount of noise cancellation that is desired. The interface circuitry may be operable to provide an interface via which a user can select from among three or more levels of noise cancellation. 117-. (canceled)18. A headset comprising:automatic noise cancellation circuitry; and the microphone is directed away from a user of the headset when the user is not speaking, and', 'the microphone is directed toward the user of the headset when the user is speaking., 'a microphone with controllable directionality, wherein19. The headset of claim 18 , wherein the automatic noise cancellation circuitry is operable to reduce a particular sound claim 18 , and wherein the particular sound is selectable.20. The headset of claim 18 , wherein the automatic noise cancellation circuitry is operable to reduce some sounds while a particular sound remains unchanged claim 18 , and wherein the particular sound that remains unchanged is selectable.21. The headset of claim 18 , wherein an amount of noise cancellation provided by the automatic noise cancellation circuitry is controllable.22. The headset of claim 18 , wherein an amount of noise cancellation provided by the automatic noise cancellation circuitry is selectable from among three or more levels of noise cancellation.23. The headset of claim 18 , wherein the headset is communicatively coupled via a wireless link to a computing device claim 18 , and wherein the automatic noise cancellation circuitry is ...

APPARATUS AND METHOD FOR SCREEN RELATED AUDIO OBJECT REMAPPING

An apparatus for generating loudspeaker signals includes an object metadata processor configured to receive metadata, to calculate a second position of the audio object depending on the first position of the audio object and on a size of a screen if the audio object is indicated in the metadata as being screen-related, to feed the first position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being not screen-related, and to feed the second position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being screen-related. The apparatus further includes an object renderer configured to receive an audio object and to generate the loudspeaker signals depending on the audio object and on position information. 1. An apparatus for generating loudspeaker signals , comprising:an object metadata processor, andan object renderer,wherein the object renderer is configured to receive an audio object,wherein the object metadata processor is configured to receive metadata, comprising an indication on whether the audio object is screen-related, and further comprising a first position of the audio object,wherein the object metadata processor is configured to calculate a second position of the audio object depending on the first position of the audio object and depending on a size of a screen if the audio object is indicated in the metadata as being screen-related,wherein the object renderer is configured to generate the loudspeaker signals depending on the audio object and depending on position information,wherein the object metadata processor is configured to feed the first position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being not screen-related, andwherein the object metadata processor is configured to feed the second position of the audio object as ...

Sound field spatial stabilizer

In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Method for determining or verifying spatial relations in a loudspeaker system

A method for determining spatial relations in a loudspeaker system comprising a central controller, multiple loudspeakers and multiple microphones. The method comprises using the multiple loudspeakers to produce sound and using the multiple microphones to measure the produced sound. Spatial relations between the loudspeakers are determined based on the measured sound, and the multiple loudspeakers are assigned to loudspeaker zones based on the determined spatial relations. Audio of the multiple loudspeakers is controlled in accordance with the assigned loudspeaker zones. Further, a method for verifying the installation of such a loudspeaker system, and a loudspeaker system with zoning and/or verification.

An Apparatus, Method and Computer Program for Audio Signal Processing

Examples of the disclosure relate to apparatus, electronic devices, methods and computer programs. The apparatus are configured to obtain at least a first audio signal and a second audio signal wherein the first audio signal and the second audio signal are captured by a microphone array including at least two microphones. The apparatus are also configured to identify at least a first direction and at least a second direction. The first and second directions are identified for a plurality of frequency bands. The first direction and the second direction are identified by using delay parameters between at least the first audio signal and the second audio signal.

Object Based Audio Rendering Using Visual Tracking of at Least One Listener

In some embodiments, a method and system for generating image data, processing the image data to generate listener data indicative of at least one listener characteristic (e.g., position and/or size of each listener), and rendering at least one audio object (e.g., rendering an object based audio program) in response to the listener data (and optionally also listener identification data). For rendering a program indicative of audio objects, at least one speaker feed may be generated for driving at least one speaker to emit sound indicative of one of the objects and additional sound indicative of another one of the objects, where the sound is intended to be perceived by a listener at a first position with balance and delay appropriate to the first position, and the additional sound is intended to be perceived by a listener at a second position with balance and delay appropriate to the second position. 1. A method for rendering an audio program comprising one or more audio objects for playback in an environment including a speaker array comprising at least one speaker , said method including the steps of:(a) generating image data indicative of at least one listener in the environment;(b) processing the image data to generate listener data indicative of at least one characteristic of at least one said listener; and(c) rendering at least one of the audio objects in response to the listener data.2. The method of claim 1 , wherein the listener data is indicative of position of at least one said listener claim 1 , and step (c) includes a step of generating at least one speaker feed for driving at least one speaker of the array to emit sound intended to be perceived by one said listener with balance and delay appropriate to the position of said listener.3. The method of claim 1 , wherein the listener data is indicative of a first position of a first listener and a second position of a second listener claim 1 , the audio program comprises at least two audio objects claim 1 , ...

An audio processing apparatus and method therefor

An audio processing apparatus comprises a receiver ( 705 ) which receives audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers ( 703 ). A renderer ( 707 ) generating audio transducer signals for the set of audio transducers from the audio data. The renderer ( 7010 ) is capable of rendering audio components in accordance with a plurality of rendering modes. A render controller ( 709 ) selects the rendering modes for the renderer ( 707 ) from the plurality of rendering modes based on the audio transducer position data. The renderer ( 707 ) can employ different rendering modes for different subsets of the set of audio transducers the render controller ( 709 ) can independently select rendering modes for each of the different subsets of the set of audio transducers ( 703 ). The render controller ( 709 ) can select the rendering mode for a first audio transducer of the set of audio transducers ( 703 ) in response to a position of the first audio transducer relative to a predetermined position for the audio transducer. The approach may provide improved adaptation, e.g. to scenarios where most speakers are at desired positions whereas a subset deviate from the desired position(s).

Adjusting Volume Levels

In general, user interfaces for controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a user interface is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network. The user interface is further configured to illustrate graphically a size of a group, the larger the group appears relatively, the more plays there are in the group. 1. A computing device , comprising:one or more processors; and displaying, via a user interface, a group volume indicator based on volume levels of at least two media players forming a media player group, a first individual volume indicator representing a volume level of a first media player, and a second individual volume indicator representing a volume level of a second media player, wherein the at least two media players comprises the first media player and the second media player, and wherein the group volume indicator is separate from the first individual volume indicator and the second individual volume indicator;', 'receiving a first input to adjust a position of the displayed first individual volume indicator of the first media player in the media player group;', 'based on receiving the first input, (i) adjusting the position of the displayed first individual volume indicator, and (ii) adjusting a position of the displayed group volume indicator;', 'receiving a second input to adjust the position of the displayed group volume indicator; and', 'based on receiving the second input, (i) adjusting the position of the displayed group volume indicator (ii) adjusting the position of the displayed first individual volume indicator based on the ...

AUGMENTED PERFORMANCE SYNCHRONIZATION

Various embodiments of the invention pertain to augmented performance synchronization systems and methods. According to some embodiments of the invention, an audio waveform may be used to generate one or more haptic waveforms for one or more electronic devices. The haptic waveforms may be generated based on any of a number of factors, including features of the audio waveform, capabilities of the haptic actuators performing the haptic waveforms, the number, type and location of devices having haptic actuators, and the like. The haptic waveforms may be synchronized with performance of the audio waveform to provide an augmented listening experience to a user. 1. A method comprising:receiving, by an electronic device including one or more speakers and a haptic actuator, an audio waveform, wherein the audio waveform is stereophonic;attenuating the audio waveform;converting the attenuated audio waveform from stereophonic to monophonic;processing the monophonic audio waveform to generate an actuator control signal;amplifying the actuator control signal;generating an audio output using the audio waveform at the one or more speakers;synchronizing transmission of the actuator control signal to the haptic actuator with transmission of the audio output to the one or more speakers; andactuating the haptic actuator with the actuator control signal while performing the audio output by the one or more speakers.2. The method of claim 1 , wherein the haptic actuator is a linear actuator.3. The method of claim 1 , wherein the electronic device further includes an input element claim 1 , and wherein the method further comprises:receiving user input from the input element; andadjusting actuation of the haptic actuator based on the user input.4. The method of claim 1 , further comprising:determining an environmental context of the electronic device; andmodifying the actuator control signal based on the environmental context.5. The method of claim 1 , further comprising:determining a ...

Facilitating Calibration of an Audio Playback Device

Example techniques facilitate calibration of a playback device. An example implementation involves a computing device capturing, via a microphone, data representing multiple iterations of a calibration sound as played by a playback device. The computing device identifies multiple sections within the captured data. Two or more sections represent respective iterations of the calibration sound as played by the playback device. Based on the multiple identified sections, the computing device determines a frequency response of the playback device, the frequency response of the playback device representing audio output by the playback device and acoustic characteristics of an environment around the playback device. Based on the frequency response of the playback device and a target frequency response, the computing device determines one or more parameters of an audio processing algorithm and sends, to the playback device, the one or more parameters of the audio processing algorithm.

Wireless communication system for use by teams

A helmet mounted communication system for use by teams includes a headset including any one of a speaker for projecting audio or a microphone for receiving audio, a mounting structure for mounting the headset in a helmet cavity of a helmet, and a biasing material between the headset and a cavity wall of the helmet cavity for floatably attaching the headset to the helmet in the helmet cavity.

Audio signal processing method for sound image localization

An audio signal processing method for sound image localization according to the present invention comprises the steps of: receiving a bit sequence including an object signal of an audio and object location information of the audio; decoding the object signal and the object location information using the received bit sequence; receiving past object location information, which is past object location information corresponding to the object location information, from a storage medium; generating an object moving path using the received past object location information and the decoded object location information; generating a variable gain value according to time using the generated object moving path; generating a corrected variable gain value using the generated variable gain value and a weighting function; and generating a channel signal from the decoded object signal using the corrected variable gain value.

BINAURAL RENDERING METHOD AND APPARATUS FOR DECODING MULTI CHANNEL AUDIO

Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal. 1. A binaural renderer in time domain , comprising:one or more processor configured to:identify an early reflection and a late reverberation for a binaural rendering;perform binaural rendering to convert a multichannel signal to a stereo signal based on the early reflection and the late reverberation.2. The binaural renderer of claim 1 , wherein the binaural rendering is performed based on binaural parameter with respect to each loudspeaker location of the multichannel signal.3. The binaural renderer of claim 1 , wherein the binaural rendering is performed by applying the late reverberating after applying the early reflection into the multichannel signal.4. The binaural renderer of claim 1 , wherein the late reverberation is extracted based on a binaural room impulse response (BRIR) for binaural rendering.5. A binaural renderer in frequency domain claim 1 , comprising:one or more processor configured to:determine an early reflection and a late reverberation for a binaural rendering;convert a multichannel audio signal to a stereo audio signal by performing binaural rendering for the multichannel audio signal,wherein the binaural rendering is performed based on early reflection and late reverberation.6. The binaural renderer of claim 5 , wherein the early reflection is processed based on bandwise partitioned convolution for binaural rendering.7. The binaural renderer of claim 5 , wherein the early reflection is determined based on a binaural room impulse responses (BRIR) in the frequency domain.8. The method of ...

Mems microphone structure and method of manufacturing the same

A MEMS microphone structure, comprising a semiconductor substrate having a cavity, a first dielectric layer having a through-hole communicating with the cavity, a lower diaphragm electrode formed above the through-hole and at least partially attached to the upper surface of the first dielectric layer, and an upper electrode structure with an insulating layer. The upper electrode structure comprises an annular supporter, a back plate having multiple holes, and an upper electrode connection. At least a part of the annular supporter extends downwardly to the lower diaphragm electrode while the rest of the annular supporter extends downwardly to the substrate. The back plate is suspended above the lower diaphragm electrode by the annular supporter, forming an air gap therebetween. An upper electrode is embedded in the insulating layer at the back plate and is lead out by the upper electrode connection.

Method of generating multi-channel audio signal and apparatus for carrying out same

A method of generating a multi-channel audio signal includes: representing locations of a plurality of speakers as a plurality of polygons whose vertices are located at locations of corresponding speakers; acquiring a location of an object sound; calculating distances between the plurality of polygons and the location of the object sound; selecting one of the plurality of polygons on the basis of the calculated distances; and generating a multi-channel audio signal that corresponds to speakers corresponding to the selected polygon by mapping the object sound to the speakers corresponding to the selected polygon.

HEADSET WITH USER CONFIGURABLE NOISE CANCELLATION vs AMBIENT NOISE PICKUP

A system comprises automatic noise cancellation circuitry and interface circuitry operable to provide an interface via which a user can configure which sounds said automatic noise cancelling circuitry attempts to cancel and which sounds said automatic noise cancelling circuitry does not attempt to cancel. The interface circuitry may be operable to provide an interface via which a user can select a sound to whitelist or blacklist. The interface circuitry may be operable to provide an interface via which a user can increase or decrease an amount of noise cancellation that is desired. The interface circuitry may be operable to provide an interface via which a user can select from among three or more levels of noise cancellation. 1. A system comprising:automatic noise cancellation circuitry in a headset, said headset comprising a microphone; andinterface circuitry operable to provide an interface via which a user can configure which sounds said automatic noise cancellation circuitry attempts to cancel and which sounds said automatic noise cancelling circuitry does not attempt to cancel, one or more of said sounds from said configuration being stored by said headset, and said microphone is configured as a noise cancellation microphone when a user of the headset is not speaking and said headset is outputting music audio.2. The system of claim 1 , wherein said interface circuitry is operable to provide said interface via which a user can select a sound that the noise cancellation circuitry attempts to cancel.3. The system of claim 1 , wherein said interface circuitry is operable to provide said interface via which a user can select a sound that the noise cancellation circuitry does not attempt to cancel.4. The system of claim 1 , wherein said interface circuitry is operable to provide said interface via which a user can increase or decrease an amount of noise cancellation that is desired.5. The system of claim 1 , wherein said interface circuitry is operable to provide ...

Audio Processing Apparatus and Method Therefor

An audio processing apparatus comprises a receiver (705) which receives audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers (703). A renderer (707) generating audio transducer signals for the set of audio transducers from the audio data. The renderer (7010) is capable of rendering audio components in accordance with a plurality of rendering modes. A render controller (709) selects the rendering modes for the renderer (707) from the plurality of rendering modes based on the audio transducer position data. The renderer (707) can employ different rendering modes for different subsets of the set of audio transducers the render controller (709) can independently select rendering modes for each of the different subsets of the set of audio transducers (703). The render controller (709) can select the rendering mode for a first audio transducer of the set of audio transducers (703) in response to a position of the first audio transducer relative to a predetermined position for the audio transducer. The approach may provide improved adaptation, e.g. to scenarios where most speakers are at desired positions whereas a subset deviate from the desired position(s).

DEVICES AND METHODS FOR CONVEYING AUDIO INFORMATION IN VEHICLES

Various methods and devices for vehicles are provided, wherein directional output of sound is used. In some embodiments, speech signals assigned to different speakers in a telephone conference are reproduced with different directional characteristics. In other embodiments, menu items of a menu are reproduced using different directions. In some embodiments, a loud-speaker arrangement may be provided at a dashboard or in a backrest of a seat of the vehicle. 1. (canceled)2. (canceled)3. (canceled)4. (canceled)5. (canceled)6. (canceled)7. (canceled)8. A method , comprising:receiving speech signals,assigning different speech signals to different speakers, andreproducing the speech signals in a vehicle with direction characteristics depending on the speaker assigned to the respective speech signal.9. The method of claim 8 , wherein the direction characteristics are generated by beam forming.10. The method of claim 8 , wherein assigning speech signals to speakers is performed based on information received together with the speech signals.11. The method of claim 8 , wherein receiving the speech signals comprises receiving the speech signals via a cellular network.12. The method of claim 8 , further comprising assigning different virtual sound sources to different speakers.13. A device claim 8 , comprising:a loudspeaker arrangement;a processing system coupled to the loudspeaker arrangement; anda receiver configured to receive speech signals, wherein the processing system is configured to assign the received speaker signals to speakers and to cause the loudspeaker arrangement to reproduce the speech signals with direction characteristics depending on the speaker.14. The device of claim 13 , wherein the receiver comprises an interface to be coupled with a mobile phone.15. (canceled)16. (canceled)17. The device of claim 13 , wherein the loudspeaker arrangement comprises at least one loudspeaker array claim 13 , said loudspeaker array comprising at least four loudspeakers ...

METHOD FOR HIGH QUALITY EFFICIENT 3D SOUND REPRODUCTION

A method for spatial sound reproduction from a first audio input signal includes using a plurality of loudspeakers using a plurality of virtual loudspeakers over which the first audio input signal is panned for forming second audio input signals using source positioning data and virtual loudspeakers positioning data. The virtual loudspeakers are synthesized by modifying second audio input signals forming third audio input signals using virtual loudspeaker spatial filter coefficients that align the loudspeakers. The method includes the steps of extracting equalization filter coefficients using source positioning data and modifying the first audio input signals using equalization filter coefficients 19. A method for spatial sound reproduction from a first audio input signal using a plurality of loudspeakers , said method using a plurality of virtual loudspeakers over which the first audio input signal is panned forming second audio input signals using source positioning data and virtual loudspeakers positioning data , said virtual loudspeakers being synthesized by modifying second audio input signals forming third audio input signals using virtual loudspeaker spatial filter coefficients () that align the loudspeakers , said method comprising the steps of:extracting equalization filter coefficients using source positioning data; and,modifying the first audio input signals using equalization filter coefficients.2. The method of claim 1 , wherein the equalization filter coefficients are retrieved from a filter database depending on the source positioning data.3. The method of claim 1 , wherein the equalization filter coefficients are calculated from an estimation of the reproduced sound field in the listening area.4. The method of claim 3 , wherein the estimation of the reproduced sound field in the listening area is performed by capturing the reproduced sound field on a plurality of microphones situated within the listening area.5. The method of claim 4 , wherein the ...

TRANSMITTING DEVICE, TRANSMITTING METHOD, RECEIVING DEVICE, AND RECEIVING METHOD

To suitably regulate sound pressure of object content on a receiving side. An audio stream including coded data of a predetermined number of pieces of object content is generated. A container of a predetermined format including the audio stream is transmitted. Information indicating a range within which sound pressure is allowed to increase and decrease for each piece of object content is inserted into a layer of the audio stream and/or a layer of the container. On a receiving side, sound pressure of each piece of object content increases and decreases within the allowable range based on the information. 1. (canceled)2. A transmission device , comprising: generate an audio stream that includes coded data of a plurality of audio objects, each of the plurality of audio objects being included in one of a plurality of content groups, and', 'generate information that indicates, for each of the plurality of content groups, (i) a designated factor type and (ii) a sound level range within which adjustment of a sound level of each audio object included in the respective content group is allowed, the sound level range being determined based on the designated factor type; and, 'circuitry configured to'}a transmitter configured to transmit the audio stream and the information.3. The transmission device according to claim 2 , wherein the circuitry is configured to insert the information into a layer of the audio stream.4. The transmission device according to claim 3 , wherein the audio stream has a coding scheme that is MPEG-H 3D Audio.5. The transmission device according to claim 3 , wherein the information indicates an upper limit value and a lower limit value of the sound level range for each of the plurality of content groups.6. The transmission device according to claim 2 , whereinthe sound level of an audio object of the plurality of audio objects is permitted to increase in response to a command that corresponds to an increase sound level instruction when the sound level ...

AUDIO RENDERING DEVICE AND AUDIO RENDERING METHOD

An audio rendering device which uses multichannel speakers includes: a first delay computation unit which computes, based on rendering information about the multichannel speakers, a first delay corresponding to a primary wavefront which propagates in a predetermined traveling direction from a sound source that is each of speakers included in the multichannel speakers; a second delay computation unit which computes, based on the rendering information about the multichannel speakers, a second delay corresponding to a secondary wavefront which is generated by the primary wavefront and has a scattering wavefront; an addition unit which adds up the first delay and the second delay to compute a total delay; and a delay filter which applies the total delay to an input audio signal to generate a multichannel audio signal for use in rendering with the multichannel speakers, and outputs the multichannel audio signal to the multichannel speakers. 113-. (canceled)14. An audio rendering device which uses multichannel speakers , the device comprising:a first delay computation unit configured to compute, based on rendering information about the multichannel speakers, a first delay which is assigned to each of speakers included in the multichannel speakers and generates a primary wavefront which propagates in a predetermined traveling direction from a sound source that is each of the speakers;a second delay computation unit configured to compute, based on the rendering information about the multichannel speakers, a second delay which is assigned to each of the speakers and generates, on the primary wavefront, a secondary wavefront having a scattering wavefront;an addition unit configured to add up the first delay and the second delay to compute a total delay for each of the speakers; anda delay filter which applies the total delay to an input audio signal to generate a multichannel audio signal for use in rendering with the multichannel speakers, and outputs the multichannel audio ...

LAYERED CODING FOR COMPRESSED SOUND OR SOUND FIELD REPRESENTENTATIONS

The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation. The method comprises sub-dividing the plurality of components into a plurality of groups of components and assigning each of the plurality of groups to a respective one of a plurality of hierarchical layers, the number of groups corresponding to the number of layers, and the plurality of layers including a base layer and one or more hierarchical enhancement layers, adding the basic side information to the base layer, and determining a plurality of portions of enhancement side information from the enhancement side information and assigning each of the plurality of portions of enhancement side information to a respective one of the plurality of layers, wherein each portion of enhancement side information includes parameters for improving a reconstructed sound representation obtainable from data included in the respective layer and any layers lower than the respective layer. The document further relates to a method of decoding a compressed sound representation of a sound or sound field, wherein the compressed sound representation is encoded in a plurality of hierarchical layers that include a base layer and one or more hierarchical enhancement layers, as well as to an encoder and a decoder for layered coding of a compressed sound representation. 1. A method of decoding a compressed Higher Order Ambisonics (HOA) representation of a sound or sound field , the method comprising: wherein the plurality of hierarchical layers have assigned thereto components of the ...

TRANSMITTING DEVICE, TRANSMITTING METHOD, RECEIVING DEVICE, AND RECEIVING METHOD FOR AUDIO STREAM INCLUDING CODED DATA

To suitably regulate sound pressure of object content on a receiving side. 1: A receiver comprising: receive an audio stream including coded data of a plurality of audio objects, each of the plurality of audio objects belongs to one of a plurality of content groups;', 'output a user interface indicating a current sound level of each of the plurality of audio objects; and', 'control a process of adjusting the sound level of each of the plurality of audio objects based on a designated factor type and sound level range information within which the sound level of the respective audio object is allowed to be adjusted for the content group to which the respective audio object belongs., 'circuitry configured to'}2: The receiver according to claim 1 , wherein the designated factor type and the sound level range information are inserted into a layer of the audio stream.3: The receiver according to claim 2 , wherein the audio stream has a coding scheme that is MPEG-H 3D Audio.4: The receiver according to claim 2 , wherein the sound level range information includes an upper limit value and a lower limit value within which the sound level is allowed to increase and decrease for each of the plurality of content groups.5: The receiver according to claim 1 , wherein the circuitry is further configured to:increase the sound level of an audio object of the plurality of audio objects when the sound level of the audio object is not at an upper limit value and when a command received is an increase sound level instruction; anddecrease the sound level of the audio object when the sound level is not at a lower limit value and when the command received is not the increase sound level instruction.6: The receiver according to claim 5 , wherein the circuitry is further configured to:decrease the sound level of another audio object of the plurality of audio objects when the command received is the increase sound level instruction; andincrease the sound level of the another audio object when ...

Method For Processing An Audio Signal, Audio Playback System, And Processing Unit For Processing Audio Signals

Method of processing an audio signal, audio reproduction system, and processing unit for conditioning audio signals. The audio signal includes at least first and second channels, the first channel being time-delayed by a predetermined delay factor, and a volume level of the first channel being attenuated by a predetermined attenuation factor. This attenuated and time-delayed first channel is added to the second channel to provide a second channel of the audio signal intended for reproduction. 1. A method of processing an audio signal comprising at least first and second channels , wherein the first channel is time-delayed by a predetermined delay factor (D) , and a volume level of the first channel is attenuated by a predetermined attenuation factor (AT) , and this attenuated and time-delayed first channel is added to the second channel to provide a second channel of the audio signal intended for reproduction.233123. The method of processing an audio signal according to claim 1 , wherein the audio signal comprises a multitude of channels which are intended for reproduction over a loudspeaker arrangement () for spatial reproduction of the audio signal claim 1 , at least first and second channels of the multitude of channels being mixed claim 1 , and the first and second channels being intended for such loudspeakers (L claim 1 , R claim 1 , LS claim 1 , RS) of the loudspeaker arrangement () which are opposite each other in relation to an axis of reflection (SA claim 1 , SA) of the loudspeaker arrangement ().3331212. The method of processing an audio signal according to claim 2 , wherein the audio signal comprises at least four audio channels for reproduction over loudspeakers (R claim 2 , L claim 2 , RS claim 2 , LS) in a loudspeaker arrangement () suitable for quadrophony or the successor technologies thereof claim 2 , wherein one loudspeaker (R claim 2 , L claim 2 , RS claim 2 , LS) each of this loudspeaker arrangement () is provided in a respective quadrant of a ...

Failure Recovery of Wearable Playback Device

An example implementation may involve a group of wearable playback devices including a first wearable playback device and a second wearable playback device. The wearable playback devices render multi-channel audio content according to particular playback responsibilities in the group. When a failure or other issue with the second wearable playback device occurs, the first playback device assumes at least part of the playback responsibility of the second playback device.

Directionality control system, calibration method, horizontal deviation angle computation method, and directionality control method

Sound collection directionality is formed toward a location corresponding to a position designated in a video of a predetermined region which is imaged by a camera apparatus with a microphone array apparatus as a reference, and audio data is collected with high accuracy. In a directionality control system ( 10 ), a signal processing unit ( 33 ) derives a sound collection direction (θ MAh ,θ MAv ) which is directed from an installation position of a microphone array apparatus ( 2 ) toward a sound position corresponding to a position designated in video data on a screen of the display device ( 36 ) in response to a user's designation of any position in the video data displayed on the screen. The signal processing unit ( 33 ) forms sound collection directionality of audio data in the derived sound collection direction (θ MAh ,θ MAv ).

WIRELESS COMMUNICATION SYSTEM FOR USE BY TEAMS

A wireless communication system comprises a first wireless communication device for a first team member, and a second wireless communication device for a second team member. The first wireless communication device comprises a microphone and a transmitter for transmitting an oral message via a first wireless communication protocol. The second wireless communication device comprises a main transceiver, a wearable audio device, and a support configured for mounting the main transceiver onto hockey shoulder pads of the second team member in a location physically remote from the wearable audio device. 1. A wireless communication system for use by members of a hockey team , the wireless communication system comprising: (i) a microphone for enabling the first team member to input an oral message; and', '(ii) a transmitter for transmitting the oral message via a first wireless communication protocol; and, '(a) a first wireless communication device for a first team member, the first wireless communication device comprising (i) a main transceiver for receiving the oral message from the first wireless communication device via the first wireless communication protocol and for retransmitting the oral message via a second wireless communication protocol;', '(ii) a wearable audio device comprising a receiver for receiving the oral message from the main transceiver via the second wireless communication protocol, and at least one speaker in communication with the receiver for generating audio corresponding to the oral message, wherein the speaker is mounted to the wearable audio device so as to be spaced apart from an ear of the second team member such that the second team member is able to hear ambient sound in addition to the oral message; and', '(iii) a support configured for mounting the main transceiver onto hockey shoulder pads of the second team member in a location physically remote from the wearable audio device;, '(b) a second wireless communication device for a second ...

APPARATUS AND METHOD FOR SCREEN RELATED AUDIO OBJECT REMAPPING

An apparatus for generating loudspeaker signals includes an object metadata processor configured to receive metadata, to calculate a second position of the audio object depending on the first position of the audio object and on a size of a screen if the audio object is indicated in the metadata as being screen-related, to feed the first position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being not screen-related, and to feed the second position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being screen-related. The apparatus further includes an object renderer configured to receive an audio object and to generate the loudspeaker signals depending on the audio object and on position information. 1. An apparatus for generating loudspeaker signals , comprising:an object metadata processor, andan object renderer,wherein the object renderer is configured to receive an audio object,wherein the object metadata processor is configured to receive metadata, comprising an indication on whether the audio object is screen-related, and further comprising a first position of the audio object,wherein the object metadata processor is configured to calculate a second position of the audio object depending on the first position of the audio object and depending on a size of a screen if the audio object is indicated in the metadata as being screen-related,wherein the object renderer is configured to generate the loudspeaker signals depending on the audio object and depending on position information,wherein the object metadata processor is configured to feed the first position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being not screen-related, andwherein the object metadata processor is configured to feed the second position of the audio object as ...

Sound field spatial stabilizer with structured noise compensation

In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Systems and Methods for Positioning a User

A hands-free intercom may include a user-tracking sensor, a directional microphone, a directional sound emitter, a display device, and/or a communication interface. The user-tracking sensor may determine a location of a user so the directional microphone can measure vocal emissions by the user and the directional sound emitter can deliver audio to the user. The hands-free intercom may provide privacy to the user. The hands-free intercom may prevent an eavesdropper from hearing the user's vocal emissions, for example, by canceling the vocal emissions at the eavesdropper's ear. The directional sound emitter may deliver out-of-phase sound to cancel the vocal emissions. The hands-free intercom may also, or instead, cancel ambient noise at the user's ear. The hands-free intercom may measure or predict a filtration of the sound to be canceled and compensate for the filtration when canceling the sound. 1. (canceled)2. A system for inducing a user to move to a predetermined location , the system comprising:one or more tracking devices that track a location of the user; anda directional sound emitter that, in response to the one or more tracking devices determining that the tracked location of the user is different from the predetermined location, delivers ultrasonic waves to the user that frequency convert at the tracked location to induce the user to move to the predetermined location.3. The system of claim 2 , wherein the ultrasonic waves frequency convert at the predetermined location to generate first audible sound waves.4. The system of claim 3 , wherein the first audible sound waves are a modified version of second audible sound waves to be delivered to the user via the directional sound emitter when the user is at the predetermined location.5. The system of claim 4 , wherein the first audible sound waves have a degraded sound quality compared to the second audible sound waves.6. The system of claim 5 , wherein the first audible sound waves have a decreased audio bit ...

ADAPTIVE BASS PROCESSING SYSTEM

An effective and simple psychoacoustic bass generation system generates a harmonic signal having inter-modulation controllable to remain below a threshold level and includes a high-pass filter configured to pass harmonics which are reproducible with fidelity by the loudspeaker or other transducer and a loudness matching block configured to compensate the loudness of the desired harmonics to match the loudness of the original signal. 1. A bass enhancement system comprising:an adaptive low frequency signal extractor configured to extract low frequency signals from an input audio signal;a nonlinear processor configured to generate from the low frequency signals harmonic signals having inter-modulation controllable to remain below a threshold level;an adaptive adjustor configured to set a gain used with the nonlinear processor for adjusting harmonic signal decay associated with the harmonic signals;a high-pass filter positioned in the output path from the nonlinear processor and configured to pass harmonic signals which are reproducible by an output transducer; andan output mixer which sums high-pass filtered input audio signals with psychoacoustic bass corresponding to the harmonic signals generated by the nonlinear processor.2. The system as recited in further comprising a loudness matching block configured to compensate the loudness of the desired harmonics to match the loudness of the low frequency signals.3. The system as recited in wherein the adaptive adjustor generates a harmonic decay rate that is implemented by an automatic gain related to at least one of an input audio level or the cut-off frequency of the output transducer.4. The system as recited in wherein the adaptive adjustor generates full-harmonic generation and inaudible inter-modulation.5. The system as recited in wherein the nonlinear processor generates both even and odd harmonics.6. The system as recited in wherein the nonlinear processor is designed as a fifth order polynomial.7. The system as ...

TRANSDUCER CONTROLLER AND METHOD THEREFOR

In an embodiment, a transducer controller is configured to apply a damping signal to reduce energy stored in the transducer after the transducer has been driven with a drive signal to form a transmitted acoustic signal. 17-. (canceled)8. A method of forming a transducer controller for an acoustic transducer comprising:configuring the transducer controller to form a transmission phase, a damping phase, and a distance measuring phase wherein the transducer controller is configured to form a drive signal to drive the acoustic transducer during the transmission phase; andconfiguring the transducer controller to, during the damping phase and prior to the distance measuring phase, apply a damping signal to the acoustic transducer wherein the damping signal forms a damping voltage that is substantially opposite in polarity to a polarity of a circulating current of the acoustic transducer.9. The method of wherein configuring the transducer controller to claim 8 , during the damping phase and prior to the distance measuring phase claim 8 , apply the damping signal includes configuring the transducer controller to receive a signal from the acoustic transducer that is representative of the circulating current and to apply the signal in response to a substantially zero crossing of the circulating current.10. The method of further including configuring the transducer controller to apply the damping signal for at least a portion of a half of a cycle of the circulating current but less than the entire half cycle.11. The method of wherein configuring the transducer controller to claim 8 , during the damping phase and prior to the distance measuring phase claim 8 , apply the damping signal includes configuring the transducer controller to apply a fixed charge to the acoustic transducer during an open-loop sub-phase of the damping phase and to apply a variable charge that is representative of residual energy stored in the acoustic transducer during a closed-loop sub-phase of the ...

BINAURAL RENDERING METHOD AND APPARATUS FOR DECODING MULTI CHANNEL AUDIO

Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal. 1. A binaural renderer in time domain , comprising:one or more processor configured to:identify an early reflection and a late reverberation for a binaural rendering;perform binaural rendering to convert a multichannel signal to a stereo signal based on the early reflection and the late reverberation.2. The binaural renderer of claim 1 , wherein the binaural rendering is performed based on binaural parameter with respect to each loudspeaker location of the multichannel signal.3. The binaural renderer of claim 1 , wherein the binaural rendering is performed by applying the late reverberating after applying the early reflection into the multichannel signal.4. The binaural renderer of claim 1 , wherein the late reverberation is extracted based on a binaural room impulse response (BRIR) for binaural rendering.5. A binaural renderer in frequency domain claim 1 , comprising:one or more processor configured to:determine an early reflection and a late reverberation for a binaural rendering;convert a multichannel audio signal to a stereo audio signal by performing binaural rendering for the multichannel audio signal,wherein the binaural rendering is performed based on early reflection and late reverberation.6. The binaural renderer of claim 5 , wherein the early reflection is processed based on bandwise partitioned convolution for binaural rendering.7. The binaural renderer of claim 5 , wherein the early reflection is determined based on a binaural room impulse responses (BRIR) in the frequency domain.8. The method of ...

WIRELESS COMMUNICATION SYSTEM FOR USE BY TEAMS

A wireless communication system comprises a first wireless communication device for a first team member, and a second wireless communication device for a second team member. The first wireless communication device comprises a microphone and a transmitter for transmitting an oral message via a first wireless communication protocol. The second wireless communication device comprises a main transceiver, and a wearable audio device. The first wireless communication device may include a push to talk hard trigger assembly. The wearable audio device may comprise an epaulet speaker. The system may include a support for mounting the main transceiver onto a shoulder pad of the second team member. 1. A wireless communication system for use by members of a team , the wireless communication system comprising: (i) a microphone for enabling the first team member to input an oral message; and', '(ii) a transmitter for transmitting the oral message via a first wireless communication protocol; and, '(a) a first wireless communication device for a first team member, the first wireless communication device comprising (i) a main transceiver for receiving the oral message from the first wireless communication device via the first wireless communication protocol and for retransmitting the oral message via a second wireless communication protocol;', '(ii) a wearable audio device comprising a receiver for receiving the oral message from the main transceiver via the second wireless communication protocol, and at least one speaker in communication with the receiver for generating audio corresponding to the oral message, wherein the speaker is mounted to the wearable audio device so as to be spaced apart from an ear of the second team member such that the second team member is able to hear ambient sound in addition to the oral message; and', '(iii) a support for mounting the main transceiver to the second team member in a location physically remote from the wearable audio device;, '(b) a ...

CONFERENCE AUDIO MANAGEMENT

A method for managing audio during a conference includes steering, at a spatial steering processor of a mobile device, first decoded audio to be projected from a speaker at a first angle. The method also includes steering, at the spatial steering processor, second decoded audio to be projected from the speaker at a second angle. The first decoded audio corresponds to a decoded version of a first audio stream from a first device associated with a first participant of the conference, and the second decoded audio corresponds to a decoded version of a second audio stream from a second device associated with a second participant of the conference. The first decoded audio is synchronized with the second decoded audio. 1. A mobile device for managing audio during a conference , the mobile device comprising: steer first decoded audio to be projected from a speaker at a first angle, the first decoded audio corresponding to a decoded version of a first audio stream from a first device associated with a first participant of the conference; and', 'steer second decoded audio to be projected from the speaker at a second angle, the second decoded audio corresponding to a decoded version of a second audio stream from a second device associated with a second participant of the conference, and the first decoded audio synchronized with the second decoded audio; and, 'a spatial steering processor configured to project the first decoded audio at the first angle; and', 'project the second decoded audio at the second angle., 'the speaker configured to2. The mobile device of claim 1 , wherein the spatial steering processor is configured to:apply a first head-related transfer function (HRTF) to the first decoded audio to steer the first decoded audio; andapply a second HRTF to the second decoded audio to steer the second decoded audio.3. The mobile device of claim 1 , wherein the first angle and the second angle are based on user-defined settings.4. The mobile device of claim 1 , wherein ...

Methods and apparatus for implementing sound events

Methods and apparatus for implementing sound events are provided. When a first sound event is triggered, a first virtual routing channel corresponding to the first sound event is determined. A link between the first virtual routing channel and a second virtual routing channel is established according to priority information of the first sound event. The second virtual routing channel is a virtual routing channel corresponding to a currently-implemented second sound event. One or more processes are performed including: using an electrical level of the first virtual routing channel to restrain an electrical level of the second virtual routing channel through the link, and using an electrical level of the second virtual routing channel to restrain an electrical level of the first virtual routing channel through the link. Accordingly, the apparatus includes a channel determining unit, a link establishing unit, and a restraining implementation unit.

Techniques for localized perceptual audio

Audio perception in local proximity to visual cues is provided. A device includes a video display, first row of audio transducers, and second row of audio transducers. The first and second rows can be vertically disposed above and below the video display. An audio transducer of the first row and an audio transducer of the second row form a column to produce, in concert, an audible signal. The perceived emanation of the audible signal is from a plane of the video display (e.g., a location of a visual cue) by weighing outputs of the audio transducers of the column. In certain embodiments, the audio transducers are spaced farther apart at a periphery for increased fidelity in a center portion of the plane and less fidelity at the periphery.

ELECTRONIC DEVICE SOUND REPRODUCTION SYSTEM

A deformable electronic device comprises a housing deformable by a user, at least one deformable speaker and a processing unit for selecting a sound reproduction method using the at least one speaker in response to deformation of the electronic device. 1. A deformable electronic device comprising:a housing deformable by a user;at least one deformable speaker; anda processing unit for selecting a sound reproduction method using the at least one speaker in response to deformation of the electronic device.2. The electronic device of claim 1 , wherein the deformation of the electronic device comprises one of claim 1 , (a) bending of the electronic device in a first front direction and (b) bending of the electronic device in a second rear direction substantially opposite to the first direction.3. The electronic device of claim 1 , wherein the sound reproduction method comprises one of monophonic and stereophonic reproduction.4. The electronic device of claim 1 , further comprising:a flexible display for outputting an image derived from a video signal received from the processing unit.5. The electronic device of claim 4 , wherein the flexible display is disposed in a front of the electronic device and the speaker is disposed in a rear of the electronic device.6. The electronic device of claim 4 , further comprising:a deformation maintenance system for maintaining the deformation of the electronic device.7. The electronic device of claim 6 , wherein the deformation maintenance system is disposed between the flexible display and the speaker claim 6 , and comprises a plurality of sequentially connected unit pieces.8. The electronic device of claim 7 , wherein a connection portion between each of the plurality of sequentially connected unit pieces suppresses rotation between the unit pieces induced by a restoring force of the electronic device that results from a device deformation.9. The electronic device of claim 7 , wherein a connection portion between the unit pieces ...

DIRECTIONALITY CONTROL SYSTEM, CALIBRATION METHOD, HORIZONTAL DEVIATION ANGLE COMPUTATION METHOD, AND DIRECTIONALITY CONTROL METHOD

A directionality control system includes: a camera; a microphone provided as a separate body from the camera; a display that displays video data captured by the camera; and a processor that computes a sound collection direction, which is directed from the microphone toward a sound position corresponding to a designated position in the video data. The processor computes the sound collection direction by using parameters including: a first height of the camera from a reference surface, a second height of the microphone from the reference surface, a third height of a computation reference point from the reference surface, the computation reference point being positioned in the sound collection direction at a position different from the sound position, a direction which is directed from the camera toward the sound position, and a fourth height of the sound position from the reference surface. 1. A directionality control system comprising;an imaging device that captures a video;a sound collector that collects sound, and being provided as a separate body from the imaging device;a display that displays video data captured by the imaging device; anda processor that computes a sound collection direction, which is directed from the sound collector toward a sound position corresponding to a designated position in the video data in response to a designation of a position in the video data displayed on the display, whereinthe processor computes the sound collection direction by using parameters including;a first height of the imaging device from a reference surface,a second height of the sound collector from the reference surface,a third height of a computation reference point from the reference surface, the computation reference point being positioned in the sound collection direction at a position different from the sound position corresponding to the designated position in the video data,a direction which is directed from the imaging device toward the sound position ...

Transmitting device, transmitting method, receiving device, and receiving method

To suitably regulate sound pressure of object content on a receiving side. An audio stream including coded data of a predetermined number of pieces of object content is generated. A container of a predetermined format including the audio stream is transmitted. Information indicating a range within which sound pressure is allowed to increase and decrease for each piece of object content is inserted into a layer of the audio stream and/or a layer of the container. On a receiving side, sound pressure of each piece of object content increases and decreases within the allowable range based on the information.

DETERMINING THE INTER-CHANNEL TIME DIFFERENCE OF A MULTI-CHANNEL AUDIO SIGNAL

A method and device are disclosed for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A determination is made at a number of consecutive time instances, inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal. Each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference. An adaptive inter-channel correlation threshold is adaptively determined based on adaptive smoothing of the inter-channel correlation in time. A current value of the inter-channel correlation is then evaluated in relation to the adaptive inter-channel correlation threshold to determine whether the corresponding current value of the inter-channel time difference is relevant. Based on the result of this evaluation, an updated value of the inter-channel time difference is determined. 1. A method for determining an inter-channel time difference of a multi-channel audio signal having at least two channels , wherein said method comprising:determining, at a number of consecutive time instances, an inter-channel correlation;evaluating a current value of inter-channel correlation to determine whether a current corresponding value of the inter-channel time difference is relevant; anddetermining an updated value of the inter-channel time difference based on the result of the evaluation.2. The method of claim 1 , wherein each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference.3. The method of claim 1 , wherein said evaluating a current value of inter-channel correlation is performed to determine whether or not the current value of the inter-channel time difference is used when determining the updated value of the inter-channel time difference.4. The method of claim 1 , wherein said determining an updated value of the inter-channel time difference ...

Updating Playback Device Configuration Information Based on Calibration Data

An example computing device is configured to perform functions including receiving calibration data corresponding respectively to a plurality of playback devices, where each playback device in the plurality of playback devices is located in a respective playback environment other than a first playback environment. The functions also include receiving playback device characteristic data respectively indicating at least one playback device characteristic for each playback device of the plurality of playback devices. The functions also include, based on at least the calibration data and the playback device characteristic data, determining updated playback device configuration information and transmitting data indicating the updated playback device configuration information to a given playback device located in the first environment. 1. A first playback device comprising:at least one processor;a non-transitory computer readable medium; and [ a first set of data that includes (i) data corresponding to a calibration event of the first playback device, (ii) data indicating configuration information for the first playback device, and (iii) data indicating at least one device characteristic of the first playback device; and', 'a second set of data that includes (i) data corresponding to a calibration event of the second playback device, (ii) data indicating configuration information for the second playback device, and (iii) data indicating at least one device characteristic of the second playback device;, 'transmit calibration data to a computing device configured to communicate with a media playback system that includes the first playback device and a second playback device, the calibration data comprising, 'receive, from the computing device, first updated configuration information for the first playback device and second updated configuration information for the second playback device;', 'apply the first updated configuration information when playing back audio content; ...

BINAURAL RENDERING METHOD AND APPARATUS FOR DECODING MULTI CHANNEL AUDIO

Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal. 1. A binaural rendering method in time domain , comprising:identifying an early reflection and a late reverberation for a binaural rendering;performing binaural rendering to convert a multichannel signal to a stereo signal based on the early reflection and the late reverberation.2. The binaural rendering method of claim 1 , wherein the binaural rendering is performed based on binaural parameter with respect to each loudspeaker location of the multichannel signal.3. The binaural rendering method of claim 1 , wherein the binaural rendering is performed by applying the late reverberating after applying the early reflection into the multichannel signal.4. The binaural rendering method of claim 1 , wherein the late reverberation is extracted based on a binaural room impulse response (BRIR) for binaural rendering.5. A binaural rendering method in frequency domain claim 1 , comprising:determining an early reflection and a late reverberation for a binaural rendering;converting a multichannel audio signal to a stereo audio signal by performing binaural rendering for the multichannel audio signal,wherein the binaural rendering is performed based on early reflection and late reverberation.6. The binaural rendering method of claim 5 , wherein the early reflection is processed based on bandwise partitioned convolution for binaural rendering.7. The binaural rendering method of claim 5 , wherein the early reflection is determined based on a binaural room impulse responses (BRIR) in the frequency domain.8. The binaural rendering ...

AUDIO NETWORK, A METHOD OF AUDIO ROUTE CONTROL IN AN AUDIO NETWORK AND A METHOD OF CONTROLLING ACCESS TO AN AUDIO SOURCE IN AN AUDIO NETWORK

An audio network including at least one audio input; at least one audio output; a controller adapted to create audio routes, each audio route comprising an audio input, an audio output and an electrical connection therebetween; a plurality of network clients, each network client being adapted to send a creation signal to the controller to create a new audio route; the controller being adapted such that on receipt of a creation signal from a second network client requesting creation of a new audio route the controller determines if creation of the new audio route involves an audio input or audio output employed in an audio route created by a first network client and to send an alert to a network client if this is the case, the controller creating the new audio route on receipt of an override signal from the network client. 1. An audio broadcast network for connection to a broadcast distribution system comprisingat least one audio input;at least one audio output for connection to the broadcast distribution system, the total number of audio inputs and outputs being greater than two;a controller adapted to create audio routes, each audio route comprising an audio input, an audio output and an electrical connection therebetween;a plurality of network clients, each network client being adapted to send a creation signal to the controller to create a new audio route;the controller being adapted such that on receipt of a creation signal from a second network client requesting creation of a new audio route the controller determines if creation of the new audio route involves an audio input or audio output employed in an audio route created by a first network client and to send an alert to a network client if this is the case, the controller creating the new audio route on receipt of an override signal from the network client.2. An audio broadcast network as claimed in claim 1 , wherein the controller sends an alert when the new audio route employs an audio output used in an ...

An Apparatus, Method and Computer Program for Obtaining Audio Signals

An apparatus, electronic device, method and computer program wherein the apparatus includes: processing circuitry; and memory circuitry including computer program code, the memory circuitry and the computer program code configured to, with the processing circuitry, enable the apparatus to perform: obtaining spatial information relating to a captured sound field from a first set of microphones; obtaining one or more signals from a second set of microphones where the one or more signals relate to the captured sound field; and using the obtained spatial information from the first set of microphones to process the one or more signals obtained from the second set of microphones; wherein the first set of microphones is provided within an electronic device and the second set of microphones is provided external to the electronic device.

Sound Masking System

A sound masking system according to the invention is disclosed in which one or more sound masking loudspeaker assemblies are coupled to one or more electronic sound masking signal generators. The loudspeaker assemblies in the system of the invention have a low directivity index and preferably emit an acoustic sound masking signal that has a sound masking spectrum specifically designed to provide superior sound masking in an open plan office. Each of the plurality of loudspeaker assemblies is oriented to provide the acoustic sound masking signal in a direct path into the predetermined area in which masking sound is needed. In addition, the sound masking system of the invention can include a remote control function by which a user can select from a plurality of stored sets of information for providing from a recipient loudspeaker assembly an acoustic sound masking signal having a-selected sound masking spectrum. 1. A sound masking system for providing sound masking in a predetermined area , said system comprising:a plurality of loudspeaker assemblies, coupled to one or more sources of an electrical sound masking signal, each comprising a reflector positioned relative to an audio emitter aperture so as to provide a low directivity index output and a face plate covering said audio emitter, the face plate having holes having a same diameter as the aperture and a total area of the holes being substantially equal to one half of an area of the face plate.2. The sound masking system of claim 1 , wherein the plurality of loudspeaker assemblies are interconnected via multi-conductor American Wire Gage (AWG) No. 24 size wiring pieces terminated at both ends with quick connect/disconnect connectors claim 1 , said quick connect/disconnect connectors corresponding to integral input and output jacks on said loudspeaker assemblies.3. The sound masking system of claim 1 , wherein claim 1 , in said plurality of loudspeaker assemblies each having a low directivity index claim 1 , each ...

Determining the inter-channel time difference of a multi-channel audio signal

A method and device are disclosed for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A determination is made at a number of consecutive time instances, inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal. Each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference. An adaptive inter-channel correlation threshold is adaptively determined based on adaptive smoothing of the inter-channel correlation in time. A current value of the inter-channel correlation is then evaluated in relation to the adaptive inter-channel correlation threshold to determine whether the corresponding current value of the inter-channel time difference is relevant. Based on the result of this evaluation, an updated value of the inter-channel time difference is determined.

Spatial Audio Apparatus

An apparatus comprising: an input configured to receive at least one audio signal from a further apparatus; an input configured to receive at least one audio signal associated with the apparatus; an orientation/location determiner configured to determine a relative orientation/location difference between the apparatus and the further apparatus; an audio processor configured to process the at least one audio signal from the further apparatus based on the relative orientation/location difference between the apparatus and the further apparatus; and a combiner configured to combine the at least one audio signal from the further apparatus having been processed and the at least one audio signal associated with the apparatus. 1. A method , comprising:determining at least one of a position and an orientation of at least one of an apparatus and at least one further apparatus wherein the apparatus is configured to record at least one audio signal from an audio source in an environment and the at least one further apparatus is configured to record the at least one audio signal in the same environment;providing a data communication between the apparatus and the at least one further apparatus to transfer the at least one audio signal from the at least one further apparatus to the apparatus based on the respective recordings in the same environment;displaying, at a display of the apparatus, an editable object based on the at least one audio signal recorded by the apparatus and displaying at least one further editable object based on the at least one audio signal recorded by the at least one further apparatus, wherein the displayed editable objects are located on the display based on the determined at least one of the position and the orientation of the at least one of the apparatus and the at least one further apparatus;moving, on the display of the apparatus, at least one of the editable object and the at least one further editable object based on an interaction with the display ...

Information processing device, information processing method, and program

The present technology relates to an information processing device, an information processing method, and a program that, when information is conveyed to a user to whom the information is desired to be conveyed, enable to convey the information to the user while the information is concealed from the other users. The information processing device includes a positional relationship detection unit that detects a positional relationship among a plurality of users, a detection unit that detects a blind spot of a first user among the plurality of users, and a display control unit that performs control for displaying information in the blind spot detected by the detection unit. The information processing device further includes a sight line detection unit that detects a sight line direction of a second user, and the display control unit performs control for displaying information in a blind spot located in the sight line direction of the second user. The present technology can be applied to an information processing device that presents information to a user.

AUDIO SPEAKER SYSTEM WITH VIRTUAL MUSIC PERFORMANCE

In a multi-speaker audio system for, e.g., a home entertainment system or other entertainment system, each networked-speaker (wired or wireless) can be assigned a particular voice, instrument, group of voices and/or instruments, or a particular stage location of a performance to reproduce a more realistic and life-like audio experience. 1. A device comprising:at least one computer readable storage medium bearing instructions executable by a processor;at least one processor configured for accessing the computer readable storage medium to execute the instructions to configure the processor for:receiving plural audio speaker identifications (IDs), each associated with a respective speaker;receiving information regarding plural tracks of an audio recording, the information indicating for each track one or more of: individual instruments, individual voice types, individual voice roles, individual instrument types, modeled stage position of a source of audio for the respective track; andmapping tracks to respective speakers.2. The device of claim 1 , wherein the processor when executing the instructions is configured for mapping tracks to respective speakers based at least in part on user input.3. The device of claim 1 , wherein the information regarding plural tracks of an audio recording is received from a storage medium bearing the audio recording.4. The device of claim 1 , wherein the information regarding plural tracks of an audio recording is received from a network server separate from a storage medium bearing the audio recording.5. The device of claim 1 , wherein the information regarding plural tracks of an audio recording indicates claim 1 , for at least one track claim 1 , an individual instrument.6. The device of claim 1 , wherein the information regarding plural tracks of an audio recording indicates claim 1 , for at least one track claim 1 , an individual voice type.7. The device of claim 1 , wherein the information regarding plural tracks of an audio ...

Method for Improving Manufacturability of Cavity Packages for Direct Top Port MEMS Microphone

A MEMS device for use in some embodiments in a microphone or pressure sensor and method of making the same wherein a portion of the package surrounding the acoustic port is deformed either away from, towards, or both away from and towards the interior of the package. By providing this raised area proximate the acoustic port, external debris is less likely to enter the acoustic port and damage the fragile MEMS die. Further, internal attachment material holding the MEMS die to the inside of the package is prevented by flowing into and obscuring the acoustic port. The advantages of this design include longer operation lifetimes for the MEMS device, greater design freedom, and increases in production yield.

User Experience Oriented Audio Signal Processing

Example embodiments disclosed herein relate to user experience oriented audio signal processing. There is provided a method for user experience oriented audio signal processing. The method includes obtaining a first audio signal from an audio sensor of an electronic device; computing, based on the first audio signal, a compensation factor for an acoustic path from the electronic device to a listener and applying the compensation factor to a second audio signal outputted from the electronic device. Corresponding system and computer program products are disclosed.

Encoding and Rendering of Object Based Audio Indicative of Game Audio Content

In some embodiments, a method (typically performed by a game console) for generating an object based audio program indicative of game audio content (audio content pertaining to play of or events in a game, and optionally also other information regarding the game), and including at least one audio object channel and at least one speaker channel. In other embodiments, a game console configured to generate such an object based audio program. Some embodiments implement object clustering in which audio content of input objects is mixed to generate at least one clustered audio object, or audio content of at least one input object is mixed with speaker channel audio. In response to the program, a spatial rendering system (e.g., external to the game console) may operate with knowledge of playback speaker configuration to generate speaker feeds indicative of a spatial mix of the program's speaker and object channel content. 127-. (canceled)28. A method for generating an object based audio program indicative of game audio content , said method including steps of:(a) determining at least one audio object channel, wherein each said audio object channel is indicative of at least one audio object, and at least one said audio object channel is indicative of game audio content;(b) determining at least one speaker channel, said at least one speaker channel being associated with a first speaker configuration, wherein at least one said speaker channel is indicative of additional game audio content; and(c) generating the object based audio program, including by encoding data indicative of each said audio object channel and each said speaker channel,wherein step (a) includes steps of:(d) identifying a set of input audio objects which are indicative of game audio content, where the set consists of N audio objects and N is a positive integer; and(e) replacing the set of input audio objects with a reduced set of audio objects, where the reduced set consists of M audio objects, M is a ...

SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD, AND PROGRAM

A signal processing device according to an embodiment includes a measurement signal generation unit that generates a first measurement signal and outputs the signal to headphones, a sound pickup signal acquisition unit that acquires a first sound pickup signal obtained by picking up the first measurement signal by a microphone, a signal analysis unit that calculates a frequency response of the first sound pickup signal, a fit determination unit that determine a fit of the headphones by comparing a response value of the frequency response in a specified frequency band with a reference value, and a display unit that outputs gradual results of the determination in accordance with a percentage of the response value with respect to the reference value. 1. A signal processing device comprising:a measurement signal generation unit configured to generate a first measurement signal and output the signal to headphones or earphones;a sound pickup signal acquisition unit configured to acquire a first sound pickup signal obtained by picking up the first measurement signal by a microphone;a signal analysis unit configured to calculate a frequency response of the first sound pickup signal;a fit determination unit configured to determine a fit of the headphones or earphones by comparing a response value of the frequency response in a specified frequency band with a reference value; andan output unit configured to output gradual results of the determination in accordance with a percentage of the response value with respect to the reference value.2. The signal processing device according to claim 1 , whereinthe measurement signal generation unit generates a second measurement signal when a fit is determined to be good by the fit determination unit,the sound pickup signal acquisition unit acquires a second sound pickup signal obtained by picking up the second measurement signal by the microphone, anda filter for out-of-head localization is generated by using the headphones or ...

VOICE OPERATED REMOTE CONTROL

A wireless device may receive from a source device an instruction to present program content. The wireless device may determine device types missing from an entertainment system based on supporting playback of the program content. A device type missing from the entertainment system may be registered. An instruction may be transmitted to an active source device to increment or decrement a counter. The counter may represent a total number of zones that receives data from the active source device. 1. A home wireless hub comprising:a wireless transceiver; anda processor; andthe wireless transceiver and the processor are configured to wirelessly register a plurality of wireless devices with the home wireless hub, wherein the plurality of wireless devices includes at least one cell phone;the processor is configured to control access of the plurality of devices, wherein the control of access includes time of day restrictions for at least one of the wireless devices; andthe wireless transceiver and the processor are configured to receive communications from the at least one cell phone, wherein the communications from the cell phone are utilized to establish the time of day restrictions.2. The home wireless hub of wherein the control of access further includes restricting access to types of content for at least one of the wireless devices.3. The home wireless hub of wherein the wireless transceiver uses WiFi.4. The home wireless hub of wherein the processor is configured to track a state of the wireless devices.5. The home wireless hub of comprising an interface for wired connection to a wired device claim 1 , wherein the processor provides access for data transfer by the wired device.6. A home wireless hub comprising:means for wirelessly register a plurality of wireless devices with the home wireless hub, wherein the plurality of wireless devices includes at least one cell phone;means for controlling access of the plurality of devices, wherein the control of access includes ...

Apparatus and method for generating a plurality of parametric audio streams and apparatus and method for generating a plurality of loudspeaker signals

An apparatus for generating a plurality of parametric audio streams from an input spatial audio signal obtained from a recording in a recording space has a segmentor and a generator. The segmentor is configured for providing at least two input segmental audio signals from the input spatial audio signal, wherein the at least two input segmental audio signals are associated with corresponding segments of the recording space. The generator is configured for generating a parametric audio stream for each of the at least two input segmental audio signals to obtain the plurality of parametric audio streams.

Concept for generating an enhanced sound field description or a modified sound field description using a multi-point sound field description

Apparatus for generating an enhanced sound field description, including: a sound field generator for generating at least one sound field description indicating a sound field with respect to at least one reference location; and a meta data generator for generating meta data relating to spatial information of the sound field, wherein the at least one sound field description and the meta data constitute the enhanced sound field description. The at least one sound field description includes in a certain case a first sound field description related to the at least one reference location and a second sound field description related to a further reference location being different from the reference location, and the meta data relating to the spatial information indicates the reference location and the further reference location or one of both reference locations and a vector extending between both reference locations.

Spatial Audio Apparatus

An apparatus comprising: an input configured to receive at least one audio signal from a further apparatus; an input configured to receive at least one audio signal associated with the apparatus; an orientation/location determiner configured to determine a relative orientation/location difference between the apparatus and the further apparatus; an audio processor configured to process the at least one audio signal from the further apparatus based on the relative orientation/location difference between the apparatus and the further apparatus; and a combiner configured to combine the at least one audio signal from the further apparatus having been processed and the at least one audio signal associated with the apparatus. 1. A method , comprising:receiving, at an apparatus, at least one microphone audio signal from a further apparatus;determining at least one of a position and an orientation of at least one of the apparatus and the further apparatus;processing, by the apparatus, the at least one microphone audio signal from the further apparatus, the processing being based on a change of the determined at least one of the position and the orientation of at least one of the apparatus and the further apparatus, the change being a movement of one of the apparatus and the further apparatus; andsynthesizing, from the processed at least one microphone audio signal based on the movement, at least one spatial audio signal;wherein the spatial audio signal reflects the change.2. The method of claim 1 , wherein receiving at least one microphone audio signal from a further apparatus comprises receiving microphone audio signals from multiple channels.3. The method of claim 1 , wherein determining at least one of a position and an orientation of at least one of the apparatus and the further apparatus comprises sensing at least one of the position and the orientation using at least one of a camera of the apparatus claim 1 , a position sensor claim 1 , and an orientation/direction ...

BINAURAL RENDERING METHOD AND APPARATUS FOR DECODING MULTI CHANNEL AUDIO

Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal. 1. A binaural rendering method in time domain , comprising:identifying an early reflection and a late reverberation for a binaural rendering;performing binaural rendering to convert a multichannel signal to a stereo signal based on the early reflection and the late reverberation.2. The binaural rendering method of claim 1 , wherein the binaural rendering is performed based on binaural parameter with respect to each loudspeaker location of the multichannel signal.3. The binaural rendering method of claim 1 , wherein the binaural rendering is performed by applying the late reverberating after applying the early reflection into the multichannel signal.4. The binaural rendering method of claim 1 , wherein the late reverberation is extracted based on a binaural room impulse response (BRIR) for binaural rendering.5. A binaural rendering method in frequency domain claim 1 , comprising:determining an early reflection and a late reverberation for a binaural rendering;converting a multichannel audio signal to a stereo audio signal by performing binaural rendering for the multichannel audio signal,wherein the binaural rendering is performed based on early reflection and late reverberation.6. The binaural rendering method of claim 5 , wherein the early reflection is processed based on bandwise partitioned convolution for binaural rendering.7. The binaural rendering method of claim 5 , wherein the early reflection is determined based on a binaural room impulse responses (BRIR) in the frequency domain.8. The binaural rendering ...

Parametric joint-coding of audio sources

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Sound system

Methods and systems for processing audio data, such as spatial audio data, which modify sound characteristics of a given component of a spatial audio signal based on a relationship between a direction characteristic of the given component and a defined range of direction characteristics. A spatial audio in a format using a spherical harmonic representation of sound components is decoded by performing a transform on the spherical harmonic representation, where the transform is based on a predefined speaker layout and a predefined rule, the predefined rule indicating a speaker gain of each speaker arranged according to the predefined layout, when reproducing sound incident form a given direction to provide alternatives to existing methods of decoding spatial audio streams, which focus on soundfield reconstruction. A plurality of matrix transforms is combined into a combined transform that is performed on an audio signal; this saves processing resources of the audio system.

METHOD AND SYSTEM FOR CREATING OBJECT-BASED AUDIO CONTENT

Described herein is a method for creating object-based audio content from a text input for use in audio books and/or audio play, the method including the steps of: a) receiving the text input; b) performing a semantic analysis of the received text input; c) synthesizing speech and effects based on one or more results of the semantic analysis to generate one or more audio objects; d) generating metadata for the one or more audio objects; and e) creating the object-based audio content including the one or more audio objects and the metadata. Described herein are further a computer-based system including one or more processors configured to perform said method and a computer program product comprising a computer-readable storage medium with instructions adapted to carry out said method when executed by a device having processing capability. 1. A method for creating object-based audio content from a text input for use in audio books and/or audio play , the method including the steps of:a) receiving the text input;b) performing a semantic analysis of the received text input;bi) determining spatial information as one or more results of the semantic analysis;c) synthesizing speech and effects based the one or more results of the semantic analysis to generate one or more audio objects;d) generating metadata for the one or more audio objects; ande) creating the object-based audio content including the one or more audio objects and the metadata.2. The method according to claim 1 , wherein the method further includes the step: bii) identifying one or more origins of speech and effects as the one or more results of the semantic analysis.3. The method according to claim 2 , wherein the origins of speech and effects include one or more of actors claim 2 , a narrator claim 2 , items and ambience.4. The method according to claim 2 , wherein the spatial information is determined for the one or more origins of speech and effects.5. The method according to claim 1 , wherein the ...

Ear model, artificial head, and measurement system and measurement method using the ear model and artificial head

This measurement system is for evaluating an acoustic apparatus ( 100 ) that includes a vibrating element ( 102 ) and allows sound to be heard via vibration transmission. The measurement system includes an ear model ( 50 ) including an artificial ear ( 51 ) modeled after a human ear and an artificial temporal bone ( 57 ) in contact with the artificial ear ( 51 ); and a vibration detection element ( 56 ) disposed in the artificial temporal bone ( 57 ).

Providing sound as originating from location of display at which corresponding text is presented

In one aspect, a device includes a processor, a display accessible to the processor, and memory accessible to the processor. The memory bears instructions executable by the processor to provide sound corresponding to a portion of text presented on the display with at least one portion of the sound being provided as if originating at least substantially from a location on the display at which the portion of text is presented on the display.

Audio system and method

Embodiments relate to, for a scene comprising a representation of at least one object and at least one sound source: obtaining a decomposition of the at least one object, the decomposition comprising at least one geometric component; modelling at least one interaction of the at least one object and the at least one sound source using the at least one geometric component; and, in dependence on the modelling of the at least one interaction, processing an audio input associated with the at least one sound source to obtain an audio output.

BIOMETRIC, PHYSIOLOGICAL OR ENVIRONMENTAL MONITORING USING A CLOSED CHAMBER

A monitoring device configured for insertion into a conduit of a subject includes a housing, at least one physiological sensor coupled to the housing, a transmitter coupled to the housing, an expandable element, inflatable element, stretched membrane or balloon coupled to the housing and configured to occlude at least a portion of the conduit, a power source attached to the housing, and a processor coupled to the housing and operatively coupled to memory containing computer instruction causing the monitoring device to obtain physiological information via the at least one physiological sensor where the physiological information includes one or more of pulse rate information, body temperature information, breathing rate information, blood pressure information, cardiac output information, or blood gas level information, and processing and analyzing the physiological information to provide a result. 1. An earphone configured to confirm the identify of a user , comprising:a first sensor generating a first set of data;a second sensor generating a second set of data;a speaker;a stent;memory;an occlusion component configured to occlude at least a portion of the ear canal, where the occlusion component is configured to fit on the stent; and obtaining the first set of data;', 'obtaining the second set of data; and', 'analyzing the first and second set of data to provide a result that comprises an identification of the user., 'a processor coupled to the memory containing computer instructions, the computer instructions when executed by the processor causes the processor to perform operations comprising;'}2. The earphone according to claim 1 , where the result identifies a user from a saved list of user's.3. The earphone according to claim 1 , further including the operation of:sending a signal to the speaker to play a pre-chosen audio prior to obtaining at least one of the first and second sets of data.4. The earphone according to claim 1 , wherein the first sensor is a ...

DETERMINING THE INTER-CHANNEL TIME DIFFERENCE OF A MULTI-CHANNEL AUDIO SIGNAL

A method and device are disclosed for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A determination is made at a number of consecutive time instances, inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal. Each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference. An adaptive inter-channel correlation threshold is adaptively determined based on adaptive smoothing of the inter-channel correlation in time. A current value of the inter-channel correlation is then evaluated in relation to the adaptive inter-channel correlation threshold to determine whether the corresponding current value of the inter-channel time difference is relevant. Based on the result of this evaluation, an updated value of the inter-channel time difference is determined. 1. A method for determining an inter-channel time difference of a multi-channel audio signal having at least two channels , wherein said method comprising:determining, at a number of consecutive time instances, an inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal;obtaining an adaptive inter-channel correlation threshold;evaluating a current value of inter-channel correlation in relation to the adaptive inter-channel correlation threshold to determine whether a current corresponding value of the inter-channel time difference is relevant; anddetermining an updated value of the inter-channel time difference based on the result of the evaluation.2. The method of claim 1 , wherein each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference.3. The method of claim 2 , wherein the obtaining adaptively determines an adaptive inter-channel correlation threshold.4. The method of claim ...

METHOD AND APPARATUS FOR AUDIO OUTPUT, AND ELECTRONIC DEVICE

Method and apparatus for audio output, and electronic device. The method applies to the electronic device including first and second speakers at left and right sides of the device placed in a horizontal state, respectively, and third and fourth speakers at left and right sides of the device placed in a vertical state, respectively, the method including: detecting placement state of the device when it is under a speaker output mode; and switching, according to the placement state of the device, a stereo output and/or a bass output of audio data to be outputted to a speaker or speakers corresponding to the placement state, and outputting the audio data in the form of stereo output and/or bass output through the speaker or speakers corresponding to the placement state. Stereo output effect can be obtained when the device is in speaker output mode, regardless of whether placed horizontally or vertically. 1. A method for audio output applicable to an electronic device , wherein the electronic device comprises a first speaker , a second speaker , a third speaker and a fourth speaker , the first speaker and the second speaker disposed at the left and right sides of the electronic device placed in a horizontal state , respectively , and the third speaker and the fourth speaker disposed at the left and right sides of the electronic device placed in a vertical state , respectively , the method comprising:detecting a placement state of the electronic device when the electronic device is under a speaker output mode; andswitching, according to the placement state of the electronic device, a stereo output and/or a bass output of audio data to be outputted to a speaker or speakers corresponding to the placement state of the electronic device, and outputting the audio data in the form of stereo output and/or bass output through the speaker or speakers corresponding to the placement state of the electronic device.2. The method according to claim 1 , wherein claim 1 ,when the ...

LAYERED CODING FOR COMPRESSED SOUND OR SOUND FIELD REPRESENTATIONS

The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation. The method comprises sub-dividing the plurality of components into a plurality of groups of components and assigning each of the plurality of groups to a respective one of a plurality of hierarchical layers, the number of groups corresponding to the number of layers, and the plurality of layers including a baselayer and one or more hierarchical enhancement layers, adding the basic side information to the base layer, and determining a plurality of portions of enhancement side information from the enhancement side information and assigning each of the plurality of portions of enhancement side information to a respective one of the plurality of layers, wherein each portion of enhancement side information includes parameters for improving a reconstructed sound representation obtainable from data included in the respective layer and any layers lower than the respective layer. The document further relates to a method of decoding a compressed sound representation of a sound or sound field, wherein the compressed sound representation is encoded in a plurality of hierarchical layers that include a base layer and one or more hierarchical enhancement layers, as well as to an encoder and a decoder for layered coding of a compressed sound representation. 1. A method of decoding a compressed Higher Order Ambisonics (HOA) representation of a sound or sound field , the method comprising:receiving a bit stream containing the compressed HOA representation corresponding to a ...

Capturing Sound

An apparatus including a body, a plurality of microphones arranged in a predetermined geometry relative to the body such that the apparatus is configured to capture sound substantially from all directions around the body to produce direction and ambience information for the captured sound, and electronics for processing signals from the plurality of microphones.

Audio Device, Audio System, and Channel Distribution Method

An audio device comprising: at least one processing circuit configured to implement stored instructions and execute a plurality of tasks, wherein the plurality of tasks include: a reading task that, from a medium that stores channel information, reads the channel information; an obtaining task that obtains individual identification information of an own device; a transmitting task that transmits the channel information read from the medium and the individual identification information to a content output device that outputs content including a plurality of channels; and a receiving task that receives at least one channel in the content.

AUDIO NETWORK, A METHOD OF AUDIO ROUTE CONTROL IN AN AUDIO NETWORK AND A METHOD OF CONTROLLING ACCESS TO AN AUDIO SOURCE IN AN AUDIO NETWORK

An audio network including at least one audio input; at least one audio output; a controller adapted to create audio routes, each audio route comprising an audio input, an audio output and an electrical connection therebetween; a plurality of network clients, each network client being adapted to send a creation signal to the controller to create a new audio route; the controller being adapted such that on receipt of a creation signal from a second network client requesting creation of a new audio route the controller determines if creation of the new audio route involves an audio input or audio output employed in an audio route created by a first network client and to send an alert to a network client if this is the case, the controller creating the new audio route on receipt of an override signal from the network client. 1. An audio network comprisingat least one audio input;at least one audio output, the total number of audio inputs and outputs being greater than two;a controller adapted to create audio routes, each audio route comprising an audio input, an audio output and an electrical connection therebetween;a plurality of network clients, each network client being adapted to send a creation signal to the controller to create a new audio route;the controller being adapted such that on receipt of a creation signal from a second network client requesting creation of a new audio route the controller determines if creation of the new audio route involves an audio input or audio output employed in an audio route created by a first network client and to send an alert to a network client if this is the case, the controller creating the new audio route on receipt of an override signal from the network client.2. An audio network as claimed in claim 1 , wherein the controller sends an alert when the new audio route employs an audio output used in an audio route created by a first network client.3. An audio network as claimed in claim 1 , wherein the controller sends the ...

Smart hub

An intelligent hub for interfacing with other devices and performing smart audio or video source selection.

ADAPTIVE AUDIO CONSTRUCTION

Described herein is a method for creating an object-based audio signal from an audio input, the audio input including one or more audio channels that are recorded to collectively define an audio scene. The one or more audio channels are captured from a respective one or more spatially separated microphones disposed in a stable spatial configuration. The method includes the steps of: a) receiving the audio input; b) performing spatial analysis on the one or more audio channels to identify one or more audio objects within the audio scene; c) determining contextual information relating to the one or more audio objects; d) defining respective audio streams including audio data relating to at least one of the identified one or more audio objects; and e) outputting an object-based audio signal including the audio streams and the contextual information. 1. A method of creating an object-based audio signal from an audio input , the audio input including one or more audio channels that are recorded to collectively define an audio scene , the one or more audio channels being captured from a respective one or more spatially separated microphones disposed in a stable spatial configuration , the method comprising:receiving, by a spatial audio analysis module, the audio input;performing spatial analysis on the one or more audio channels to identify one or more audio objects within the audio scene;determining contextual information relating to the one or more audio objects, wherein the contextual information includes spatial properties of the one or more audio objects;defining respective audio streams including audio data relating to at least one of the identified one or more audio objects;outputting, by the spatial audio analysis module, an object-based audio signal including the audio streams and the contextual information;receiving, by an automated effects processing module, the object-based audio signal;performing effects processing on one or more of the audio streams to ...

RACE PROGRESS APPARATUS

A portable unit usable by a person when carrying out an exercise is described. The unit is programmable with details of a virtual athlete and their performance in a specific race, to enable the user to race against the virtual athlete. The unit informs the user through headphones of the progress of the virtual athlete during a race being carried out by a user. The unit may also be programmable with periodic encouragement messages for a user. 126-. (canceled)27. A portable device comprising:a set of earphones wearable by a person when carrying out exercise, wherein the set of earphones comprises;a casing;a programmable unit incorporated into the casing, the programmable unit being programmable to provide messages of time elapsed at any point during a workout;a connection port for a wired connection, or a component to provide a wireless connection, to connect to a separate sound unit to the earphones, wherein the separate sound unit provides music to the earphones which is played to the user during the workout;wherein the programmable unit is configured to interrupt the playing of music from the separate sound unit when the programmable unit provides a message of time elapsed.28. The portable device of claim 27 , wherein the separate sound player comprises an MP3 player.29. The portable device of claim 27 , wherein the casing and the earphones are waterproof so that the apparatus is usable by a person swimming.30. The portable device of claim 27 , wherein the connection port receives a wire from the separate sound player claim 27 , wherein the music is transferred from the separate sound player to the earphones via the wire.31. The portable device of claim 27 , wherein the connection port creates a wireless connection between the earphones and the separate sound player claim 27 , wherein the music is transferred from the separate sound player to the earphones through the wireless connection. This invention concerns a race progress apparatus.A number of race progress ...

Adjusting Volume Levels

In general, user interfaces for controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a user interface is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network. The user interface is further configured to illustrate graphically a size of a group, the larger the group appears relatively, the more plays there are in the group. 1. A computer-implemented method , comprising:providing, by a computing device, a group volume control and a plurality of individual volume controls, wherein each individual volume control includes a setting to control a volume of a respective individual audio output device and the group volume control including a setting for controlling at least two of the plurality of individual volume controls, each having an initial relationship with the group volume control; and 'adjusting a setting of at least one individual volume control of the plurality of individual volume controls in proportion to the adjustment of the setting of the group volume control.', 'in response to receiving the adjustment, 'receiving an adjustment of a setting of a volume control, wherein2. The method of claim 1 , wherein adjusting the setting of the at least one individual volume control of the plurality of individual volume controls in proportion to the adjustment of the setting of the group volume control comprises:adjusting the setting of the at least one individual volume control in proportion to the adjustment of the setting of the group volume control to preserve a relative volume difference between the at least one individual volume control and the group volume control.3. ...

PASSIVE AND ACTIVE VIRTUAL HEIGHT FILTER SYSTEMS FOR UPWARD FIRING DRIVERS

Embodiments are directed to a virtual height filter for use with or in an upward-firing speaker system that reflects sound off a ceiling to a listening location at a distance from a speaker, and that provides height cues to reproduce audio objects that have overhead audio components. A virtual height filter based on a directional hearing model is applied to the upward-firing driver signal to improve the perception of height for audio signals transmitted by the virtual height speaker to provide optimum reproduction of the overhead reflected sound. The virtual height filter is provided by any one or combination of analog or digital filter circuits, or mechanical structures including speaker grill, enclosure, or driver design or configuration. 1. An apparatus comprising:an interface to a speaker system having at least an upward-firing driver transmitting sound waves to be reflected off of an upper surface of a listening environment through a projection angle angled upward relative to a projection angle of a direct-firing driver;a virtual height filter applying a frequency response curve to a signal transmitted to the upward-firing driver to create a target transfer curve that imparts a frequency response to the reflected sound waves that accentuates a perception of virtual height to a listener in the listening environment; anda crossover coupled to the virtual height filter having a low-pass section configured to transmit low frequency signals to a direct-firing driver and a high-pass section configured to transmit high frequency signals above to the upward-firing driver.2. The apparatus of claim 1 , wherein the upper surface comprises a ceiling above the apparatus claim 1 , and the virtual height filter compensates for height cues present in sound waves transmitted directly through the listening environment in favor of height cues present in the reflected sound waves projected off the upper surface of the listening environment.3. The apparatus of claim 1 , wherein the ...

Automatic microphone switching

A method comprising: initiating a sound capture event within an apparatus; receiving at least two audio signals during the sound capture event, wherein the at least two audio signals are provided by at least two microphones associated with the apparatus; determining at least one microphone operational parameter based on the received at least two audio signals; and controlling the sound capture event such that at least one of the at least two audio signals is processed based on the at least one microphone operational parameter.