КОДИРОВАНИЕ СТЕРЕОФОНИЧЕСКИХ СИГНАЛОВ

Изобретение относится к кодированию многоканальных сигналов, более конкретно к кодированию полифонических звуковых сигналов, таких как стереофонические сигналы. Описан способ кодирования многоканального сигнала, такого как стереофонический звуковой сигнал, включающего в себя по меньшей мере первый компонент сигнала (L) и второй компонент сигнала (R). Способ содержит этапы преобразования по меньшей мере первого и второго компонентов сигнала посредством заранее заданного преобразования в главный сигнал (у), включающий в себя большую часть энергии сигнала, и по меньшей мере один остаточный сигнал (r), включающий в себя меньше энергии, чем главный сигнал, причем это заранее заданное преобразование параметризовано по меньшей мере одним параметром преобразования; и представления многоканального сигнала по меньшей мере посредством главного сигнала и параметра преобразования. Дополнительно описаны соответствующее устройство для кодирования многоканальных сигналов и соответствующий способ и устройство ...

ПАРАМЕТРИЧЕСКАЯ РЕКОНСТРУКЦИЯ АУДИОСИГНАЛОВ

Изобретение относится к средствам для параметрической реконструкции аудиосигналов. Технический результат заключается в повышении эффективности кодирования многоканального аудиосигнала. Система кодирования кодирует -канальный аудиосигнал ( ), где , в качестве одноканального сигнала понижающего микширования ( ) вместе с параметрами сухого и влажного повышающего микширования ( ). В системе декодирования секция декорреляции выводит канальный декоррелированный сигнал ( на основании сигнала понижающего микширования; при этом секция сухого повышающего микширования линейно отображает сигнал понижающего микширования в соответствии с коэффициентами сухого повышающего микширования ( ), определенными на основании параметров сухого повышающего микширования; при этом секция влажного повышающего микширования заполняет вспомогательную матрицу и линейно отображает декоррелированный сигнал в соответствии с коэффициентами влажного повышающего микширования. 5 н. и 17 з.п. ф-лы, 16 ил.

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

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

ПРОЦЕССОР АУДИОСИГНАЛА, СИСТЕМА И СПОСОБЫ РАСПРЕДЕЛЕНИЯ ОКРУЖАЮЩЕГО СИГНАЛА ПО МНОЖЕСТВУ КАНАЛОВ ОКРУЖАЮЩЕГО СИГНАЛА

Изобретение относится к средствам для распределения сигнала по множеству каналов. Технический результат заключается в повышении эффективности получения окружающих сигналов. Получают каналы окружающего сигнала таким образом, что компоненты окружающего сигнала распределяются по каналам окружающего сигнала в зависимости от положений или направлений источников звука во входном аудиосигнале. Количество полученных каналов окружающего сигнала, содержащих разный аудиоконтент, больше, чем количество каналов входного аудиосигнала. Получают прямой сигнал, который содержит компоненты прямого звука, на основе входного аудиосигнала. Выделяют окружающий сигнал на основе входного аудиосигнала. Распределяют окружающий сигнал по множеству каналов окружающего сигнала в зависимости от положений или направлений источников звука во входном аудиосигнале, при этом количество каналов окружающего сигнала больше, чем количество каналов входного аудиосигнала. Каналы окружающего сигнала и каналы прямого сигнала ассоциированы ...

ПОСЕГМЕНТНАЯ НАСТРОЙКА ПРОСТРАНСТВЕННОГО АУДИОСИГНАЛА К ДРУГОЙ УСТАНОВКЕ ГРОМКОГОВОРИТЕЛЯ ДЛЯ ВОСПРОИЗВЕДЕНИЯ

Изобретение относится к средствам посегментной настройки пространственного аудиосигнала к другой установке громкоговорителей для воспроизведения. Технический результат заключается в сохранении пространственного образа аудиосцены при перенастройке аудиосигнала к другой установке громкоговорителей. Предложено устройство для адаптации пространственного аудиосигнала, предназначенного для исходной установки громкоговорителя, к установке громкоговорителя для воспроизведения, которая отличается от исходной установки громкоговорителя. Устройство содержит блок декомпозиции на прямой звук и звук окружения, который выполнен с возможностью осуществления декомпозиции канальных сигналов в сегменте исходной установки громкоговорителя на компоненты прямого звука и окружения, и определения направления прихода для компонентов прямого звука. Блок представления прямого звука принимает информацию установки громкоговорителя для воспроизведения и настраивает компоненты прямого звука с использованием информации ...

УСТРОЙСТВО И СПОСОБ ДЛЯ КОДИРОВАНИЯ МНОГОКАНАЛЬНОГО ЗВУКОВОГО СИГНАЛА

Изобретение относится к кодированию многоканального звукового сигнала, в частности к сведению фонограмм стереофонического речевого сигнала к монофоническому сигналу для кодирования с помощью монофонического кодера, такого как кодер линейного предсказания. Техническим результатом является повышение качества и эффективности кодирования. Указанный результат достигается тем, что устройство для кодирования многоканального звукового сигнала содержит приемник многоканального звукового сигнала, содержащего первый и второй звуковые сигналы от первого и второго микрофонов, модуль разности времени для определения межвременной разности между первым и вторым звуковыми сигналами посредством объединения последовательных наблюдений взаимных корреляций между первым и вторым звуковыми сигналами, при этом взаимные корреляции нормализуются для вывода вероятностей состояния, накапливаемых с использованием алгоритма Витерби, для достижения межвременной разности со встроенным гистерезисом, и алгоритм Витерби ...

СПОСОБ И УСТРОЙСТВО ДЛЯ ОБРАБОТКИ ЗВУКОВОГО СИГНАЛА

Изобретение относится к устройству и способу обработки звукового сигнала, а именно, предназначено для обработки звукового сигнала, принятого с цифрового носителя, и широкополосного сигнала. Способ обработки звукового сигнала включает в себя прием смешанного сигнала, включающего в себя, по меньшей мере, один сигнал объекта, и информации об объекте, извлеченной при формировании смешанного сигнала, прием информации о сведении для управления сигналом объекта, формирование одной из информации о смешивании и многоканальной информации с использованием информации об объекте и информации о сведении в соответствии с режимом вывода, и если формируется информация о смешивании, формирование выходного сигнала путем применения информации о смешивании к смешанному сигналу, где смешанный сигнал и выходной сигнал соответствуют монофоническому сигналу, и где многоканальная информация соответствует информации для разложения смешанного сигнала на множество канальных сигналов. Технический результат - предотвратить ...

СПОСОБ ТРАНСЛЯЦИИ СТЕРЕОФОНИЧЕСКОГО СИГНАЛА

Изобретение относится к области радиовещания и звукозаписи стереофонических сигналов. Технический результат - уменьшение объема анализа, сокращение количества необходимой аппаратуры, сокращение потока передаваемых данных. Технический результат достигается за счет того, что в способе трансляции стереофонического сигнала на передающей стороне путем анализа формы текущего спектра сигналов левого и правого каналов формируют дополнительные сигналы, которые кодируют и передают с основным сигналом, а на приемной стороне разделяют основной и дополнительные сигналы и производят фильтрацию основного сигнала фильтрами, характеристики которых управляются декодированными дополнительными сигналами, причем в качестве основного используют монофонический сигнал. 2 ил.

УСТРОЙСТВО ПРЕДОСТАВЛЕНИЯ АУДИО И СПОСОБ ПРЕДОСТАВЛЕНИЯ АУДИО

Изобретение относится к обработке аудиосигналов. Технический результат – оптимизация канального аудиосигнала для среды прослушивания. Устройство предоставления аудио включает в себя блок приема, блок рендеринга, выполненный с возможностью устранения разности фаз между коррелируемыми входными канальными сигналами из числа множества входных канальных сигналов и преобразования множества входных канальных сигналов, в том числе коррелируемых входных канальных сигналов, во множество выходных канальных сигналов на основе схемы ввода, схемы вывода и информации для определения, выполнять ли 3D-рендеринг или 2D-рендеринг. 3 н. и 2 з.п. ф-лы, 18 ил.

УСТРОЙСТВО ПРЕДОСТАВЛЕНИЯ АУДИО И СПОСОБ ПРЕДОСТАВЛЕНИЯ АУДИО

Изобретение относится к обработке аудиосигналов. Технический результат – оптимизация канального аудиосигнала для среды прослушивания. Устройство предоставления аудио включает в себя блок рендеринга объекта, который выполняет рендеринг объектного аудиосигнала посредством использования геометрической информации, касающейся объектного аудиосигнала, блок рендеринга канала, который выполняет рендеринг аудиосигнала, имеющего первое количество каналов, в аудиосигнал, имеющий второе количество каналов, и блок смешивания, который смешивает рендерируемый объектный аудиосигнал с аудиосигналом, имеющим второе количество каналов. 3 н. и 6 з.п. ф-лы, 18 ил.

ПОСЕГМЕНТНАЯ НАСТРОЙКА ПРОСТРАНСТВЕННОГО АУДИОСИГНАЛА К ДРУГОЙ УСТАНОВКЕ ГРОМКОГОВОРИТЕЛЯ ДЛЯ ВОСПРОИЗВЕДЕНИЯ

КОМПОНОВКА ГРОМКОГОВОРИТЕЛЕЙ ДЛЯ ВОСПРОИЗВЕДЕНИЯ ТРЕХМЕРНОГО ЗВУКА В АВТОМОБИЛЯХ

АКУСТИЧЕСКАЯ СИСТЕМА

... 1. Акустическая система, содержащая источник звукового сигнала, усилитель и электроакустический преобразователь в виде резонатора, включающего корпус, ряд головок громкоговорителей, связанных с резонатором посредством опорных колец и разнесенных вдоль оси резонатора, акустический отражатель, выполненный в виде диска, который установлен оппозитно диффузорам громкоговорителей вдоль оси резонатора посредством стержней, отличающаяся тем, что она снабжена приспособлением для настройки резонанса корпуса и дополнительными акустическими отражателями, при этом последние выполнены в виде опорных колец, связанных со стержнями подвижно в направлении оси резонатора.2. Акустическая система по п.1, отличающаяся тем, что приспособление для настройки резонанса корпуса выполнено в виде винтовых пар и связано со стержнями и дополнительными акустическими отражателями.

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

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

РЕЗОНАНСНАЯ АКУСТИЧЕСКАЯ СИСТЕМА

... 1. Резонансная акустическая система, содержащая источник звукового сигнала, усилитель и электроакустический преобразователь в виде резонатора, включающего корпус, ряд головок громкоговорителей, разнесенных вдоль оси резонатора, отличающаяся тем, что она снабжена акустическими отражателями, выполненными в виде дисков, которые установлены оппозитно диффузорам громкоговорителей вдоль оси резонатора, при этом один громкоговоритель и соответствующий ему отражатель выбраны в качестве базовых, а остальные громкоговорители удалены от базового отражателя на расстояние h, определяемое величиной hm = c/(2•fn•m±1/2), где fn - частоты музыкального звукоряда; C - скорость звука в среде; m = 1,2,3 ... k - целые числа, равные порядковому номеру громкоговорителя, корпус резонатора в раскрыве связан с одним из отражателей, характерные размеры корпуса пропорциональны длине волны основного тона, а сам корпус выполнен из акустически активного материала. 2. Резонансная акустическая система по п.1, отличающаяся ...

VORRICHTUNG ZUR ERZEUGUNG EINES NACHHALLS BEI EINEM SCHALLSIGNAL

SOUND REPRODUCING SYSTEM HAVING SONIC IMAGE LOCALIZATION NETWORKS

System für Raumklangeffekt und Verfahren dafür

SECHSACHSIGER RAUMKLANGPROZESSOR MIT AUTOMATISCHER BALANCEREGLUNG UND KALIBRIERUNG

Elektroakustische UEbertragungsvorrichtung

VARIABLE MATRIX DECODER

Sound imaging system for a video game

Audio enhancement module for portable media player

An audio or video enhancement module incorporates advanced audio enhancement effects. These advanced effects cover a wide range including but not limited to 3D audio spatialization (virtualization) and transient enhancement. The device is configured for receiving an audio input signal, in analog or digital form, enhancing the received signal, and transmitting an output audio signal having the enhanced effects embedded in the signal. The output signal may be either digital or audio. Further, the output signal may be a mono signal, a stereo signal, a MIDI signal, or a mulitchannel signal (such as 2.1, 5.1, 7.1, etc.). User input to the module is provided to control the amount of audio enhancement.

Producing simulated sound distance effects

Audio signal processing means produces an output 24 having an illusory distance effect for a sound source signal S by feeding it via a direct signal path 25 and an indirect signal path 22, 23 passing through early reflection simulation means 1 which feed an output mixing means 9. Control means adjust the relative delays 3, 4 and relative gains 5, 6 in the direct 25 and indirect 22, 23 signal paths to modify the illusory distance effect so as to substantially maintain the mathematical relationship between the gains and time delays of simulated reflections relative to first sound arrivals at the output 24 encountered for sounds at that source distance in actual rooms. Signal paths 22, 23, 24, 25 may be stereophonic or multichannel using matrix gain coefficients in the early reflection simulator 1, and may produce different simulated distances for different sound positions. A plurality of sound sources S may have different simulated distances while feeding a common early reflection simulator ...

REVERBERATION APPARATUS

AUDIO CHANNEL TRANSLATION

REAL TIME AUDIO THREE-DIMENSIONAL SOUND SYSTEM WITH CONTROL ON HIGH LEVEL

ANORDNUNG ZUR DRAHTLOSEN UBERTRAGUNG VON NIEDERFREQUENZSIGNALEN

ADAPTIVE CONTROLLING OF AN ECHO WAKE FOR THE PSEUDO STEREO AUDIO SYNTHESIS.

PRODUCTION OF DEKORRELIERTER SIGNALS

DEVICE AND PROCEDURE FOR SYNTHESIZING THREE OUTPUT CHANNELS USING TWO INPUT PORTS

PROCEDURE AND DEVICE FOR THE PROCESSING OF AUDIO SIGNALS

PROCEDURE AND ARRANGEMENT FOR A DECODER FOR MEHRKANAL SURROUNDTON

TO CODING OF SOUND SIGNALS WITH RESOUND

PROCEDURE AND DEVICE FOR THE PARAMETRIC ONE STEREO AUDIODEKODIERUNG

MECHANISM FOR THE PRODUCTION A MULTI-CHANNEL - OF A SOUND SIGNAL

ELECTROACOUSTIC TRANSFORMATION WITH DEVICES FOR THE IMPROVEMENT OF THE LOW FREQUENCIES

WITH PHASE ALLOCATION OF WORKING, STEREOPHONER SCHALLSYNTHESIZER.

MULTI-PATH MORE AUDIODEKODER

RECORDING AND RENDITION OF A TWO-CANAL SYSTEM FOR GETTING OFTEN NICHE TONWIEDERGABE

MECHANISM FOR THE PRODUCTION A MULTI-CHANNEL - OF A SOUND SIGNAL

PROCEDURE AND ARRANGEMENT FOR THE PRODUCTION OF A SECOND AUDIO SIGNAL OF A FIRST AUDIO SIGNAL

SOUND IMAGING METHOD AND APPARATUS

Apparatus and method for generating an ambient signal from an audio signal, apparatus and method for deriving a multi-channel audio signal from an audio signal and computer program

REVERBERATION APPARATUS

Audio distributor

SOUND REPRODUCTION SYSTEM, PROGRAM AND DATA CARRIER

Apparatus and method for processing stereo signals for reproduction in cars to achieve individual three-dimensional sound by frontal loudspeakers

Apparatus and method for processing stereo signals for reproduction in cars to achieve individual three-dimensional sound by frontal loudspeakers Embodiments provide a digital processor (100) comprising an ambient portion extractor (102) and a spatial effect processing stage (104). The ambient portion extractor (102) is configured to extract an ambient portion from a multi-channel signal. The spatial effect processing stage (104) is configured to generate a spatial effect signal based on the ambient portion of the multi-channel signal. The digital processor (100) is configured to combine (116) the multi-channel signal or a processed version thereof with the spatial effect signal.

Method and apparatus for providing customised sound distributions

A speaker system is disclosed for providing customised acoustical wavefronts with vertical and horizontal pattern control and amplitude and phase control. The system including a speaker housing (1) having therein at least a first array (2) of high frequency driver segments (3) and at least a secondary array (4) of low frequency driver segments (5) disposed behind said first array (2), said first array having sufficient space between said driver segments (3) to allow acoustic transparency whereby a wavefront from said secondary array (4) can substantially pass through said first array (2).

SURROUND-SOUND SYSTEM

Method for deriving at least three audio signals from two input audio signals

STEREOPHONIC VOICE SIGNAL TRANSMISSION SYSTEM

SYNTHESIS OF SIGNALS FOR IMMERSIVE AUDIO PLAYBACK

A method for synthesizing sound includes receiving one or more first inputs (80), each including a respective monaural audio track (82). One or more second inputs are received, indicating respective three-dimensional (3D) source locations having azimuth and elevation coordinates to be associated with the first inputs. Each of the first inputs is assigned respective left and right filter responses based on filter response functions that depend upon the azimuth and elevation coordinates of the respective 3D source locations. Left and right stereo output signals (94) are synthesized by applying the respective left and right filter responses to the first inputs.

AUDIO SIGNAL PROCESSING SYSTEM

RECONSTRUCTING AUDIO SIGNALS WITH MULTIPLE DECORRELATION TECHNIQUES

Systems and methods of audio signal processing are provided that relate to improved upmixing, whereby N audio channels are derived from M audio channels, a decorrelated version of the M audio channels and a set of spatial parameters. The set of spatial parameters includes an amplitude parameter and a correlation parameter. The M audio channels are decorrelated using multiple decorrelation techniques to obtain the decorrelated version of the M audio channels. This can be used, for example, for generating an N audio channel upmix.

MULTICHANNEL AUDIO CODING

... ²Systems and methods of audio signal processing are provided that relate to ²improved upmixing, whereby N audio channels are derived from M audio channels ²and ²spatial parameters. The solution involves introducing uncorrelated signal ²components into ²correlated upmix signal components in response to spatial parameters which are ²interpolated ²over time. This can be used, for example, for generating an N channel upmix ²from M channels ²and spatial parameters.²² ...

SOUND SYSTEM

A calculation unit for a sound system comprises input means, a processor and output means. The input means have the purpose to receive an audio stream to be reproduced using the sound system. The output means has the purpose to control the sound system based on a first and a second plurality of individual audio signals. The processor is configured to calculate the first plurality of audio signals such that beamforming is performed by the array and to calculate the second plurality of individual audio signals to perform, using the sound system, direct sound suppression such that sound is canceled towards a listening direction. Furthermore, the processor filters at least the second plurality using a second passband characteristic comprising a second portion of the entire frequency range.

APPARATUS AND METHOD FOR PROCESSING STEREO SIGNALS FOR REPRODUCTION IN CARS TO ACHIEVE INDIVIDUAL THREE-DIMENSIONAL SOUND BY FRONTAL LOUDSPEAKERS

Apparatus and method for processing stereo signals for reproduction in cars to achieve individual three-dimensional sound by frontal loudspeakers Embodiments provide a digital processor (100) comprising an ambient portion extractor (102) and a spatial effect processing stage (104). The ambient portion extractor (102) is configured to extract an ambient portion from a multi-channel signal. The spatial effect processing stage (104) is configured to generate a spatial effect signal based on the ambient portion of the multi-channel signal. The digital processor (100) is configured to combine (116) the multi-channel signal or a processed version thereof with the spatial effect signal.

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

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

SOUND IMAGING APPARATUS FOR A VIDEO GAME SYSTEM

... 38356 SUMMARY OF THE INVENTION The game software (10) for a video game system includes sound positioning information that utilizes two spaced-apart speakers (26, 28) to give the player the impression that the sound is emanating from a location other than the actual speaker locations, for example, to the far right or the far left of the speakers. A sound processor (220) operates upon monaural signals from an audio synthesizer (180) in response to the sound positioning information from the game software (10) so that each monaural signal from the audio synthesizer is divided into two signals and at least one of the signals is passed through a transfer function to produce twochannel output signals that have a differential phase and amplitude relationship therebetween that is adjusted on a frequency dependent basis. Each different sound location generally requires a specific different relationship, and the sound processor for the video game may include a number of different transfer functions ...

Vorrichtung zur Übertragung von elektrischen Schwingungen akustischer Frequenz.

Autonomous [...]supplementary signals[...] and recovery of.

Die Erfindung betrifft ein Verfahren zum Dekodieren oder Upmixen eines Upmixsignals mit zumindest fünf Kanälen aus einem Downmixsignal mit zumindest drei Kanälen. Das Verfahren umfasst die Schritte: Durchführen zumindest zweier Korrelationsvergleiche zwischen zumindest zwei Kanalpaaren der zumindest drei Kanäle des Downmixsignals, wobei jeder Korrelationsvergleich eines der zumindest zwei Kanalpaare zumindest ein Korrelationsvergleichssignal ergibt, wobei das zumindest eine Korrelationsvergleichssignal eines der zumindest zwei Kanalpaare ein gemeinsames Signal eines der Kanalpaare und/oder ein erstes individuelles Signal eines der Kanalpaare und/oder ein zweites individuelles Signal eines der Kanalpaare aufweist, wobei das gemeinsame Signal eines der zumindest zwei Kanalpaare die gemeinsamen Signalanteile dieses Kanalpaars enthält und/oder das erste individuelle Signal eines der zumindest zwei Kanalpaare, die dem ersten Kanal des Kanalpaars spezifischen Signalanteile enthält und/oder das ...

Method and apparatus for generating a [...][...][...] from a.

Zur Vermeidung von Residualen in Fourier-transformierten Signalen werden in der erfindungsgemässen Downmixvorrichtung, die eine Fouriertransformationsvorrichtung und einen Downmixer (52) umfasst, für unterschiedliche Kanäle unterschiedliche Fensterfunktionen verwendet. In der Upmix- oder Kodiervorrichtung werden die Kanäle des Downmix durch Korrelationsvergleich in zwei individuelle Signale und ein korreliertes Signal aufgeteilt und die drei Kanäle des Upmixsignals durch geschickte Kombination der zwei Kanäle des Downmix und der beiden individuellen Signale und des korrelierten Signals erhalten.

Device and procedure for the evaluation and optimization of signals on the basis of algebraic invariants.

Die Erfindung betrifft ein Verfahren zur Evaluierung einer Verknüpfung von zwei oder mehreren Signalen (5003, 5005) auf der reellen bzw. kompletten Zahlenebene. Für gleich lange Signalabschnitte werden Abbildungen der Verknüpfungen betrachtet und für diese Abbildungen algebraische Invarianten bestimmt. Diese Invarianten werden anschliessend als «tags» benutzt, um beispielsweise eine konvergente Gewichtsfunktion zu definieren. Eine Anwendung ist insbesondere die stochastische Betrachtung von Audiosignalen, wie sie etwa aus dem Bereich Digital Audio Broadcasting bekannt ist.

Device and procedure for the temporal evaluation and optimization of stereophonen or pseudostereophonen signals.

Die vorliegende Erfindung betrifft eine Vorrichtung und ein Verfahren zur Stereophonisierung eines Monosignals (M). Laufzeitdifferenzen werden mit einem optimierten zeitlichen Parameter (S) multipliziert. Resultat ist ein Stereosignal (L,R).

Apparatus for production of pseudo-stereophonic signals based on frequency modulated stereo signals, comprises circuit with stereo converter for pseudo-stereo conversion, where two subsequent panoramic potentiometers are configured

The apparatus comprises a circuit with a stereo converter for pseudo-stereo conversion, where two subsequent panoramic potentiometers are configured. Each panoramic potentiometer forms two busbar signals. The circuit is alternatively equipped with a stereo converter and a stereo amplifier for amplifying an input signal of the stereo converter. An independent claim is also included for a method for production of pseudo-stereophonic signals with variable degree of correlation between left and right signals.

Speaker for reproducing surround sound

Spatial upmix controlled by renderer

Method for deriving at least three audio signals from two input audio signals

Method for implementing surround sound system using a plurality of vehicles

DECODEUR MATRICIEL VARIABLE

DECODEUR POUR LE TRAITEMENT D'AU MOINS DEUX SIGNAUX DE CANAL DANS UN SYSTEME QUADRIPHONIQUE DANS LEQUEL AU MOINS QUATRE SIGNAUX D'ENTREE CONTENANT DES INFORMATIONS DIRECTIONNELLES SONT CODEES EN AU MOINS DEUX SIGNAUX DE CANAL, CE DECODEUR COMPRENANT DES CONVERTISSEURS LOGARITHMIQUES DIFFERENTIELS 22, 24, POUR PRODUIRE DES SIGNAUX DE DOMINANCE INDIQUANT, RESPECTIVEMENT, LE RAPPORT DES AMPLITUDES DE DEUX SIGNAUX D'ENTREE ET LE RAPPORT DES AMPLITUDES DE LA SOMME ET DE LA DIFFERENCE DE CES SIGNAUX D'ENTREE, ET DES MOYENS MATRICIELS 100 RECEVANT LES DEUX SIGNAUX DE CANAL ET LES DEUX SIGNAUX DE DOMINANCE POUR PRODUIRE QUATRE SIGNAUX DE SORTIE L, C, R, S DONT LES EFFETS DIRECTIONNELS SONT AMELIORES. L'INVENTION S'APPLIQUE AUX SYSTEMES DE REPRODUCTION SONORE QUADRIPHONIQUES.

DISPOSITIF DE RESTITUTION D'UN SON STEREOPHONIQUE OU PSEUDO-STEREOPHONIQUE A PLUSIEURS AUDITEURS ET ENSEMBLE STEREOPHONIQUE OU PSEUDO-STEREOPHONIQUE COMPORTANT UN TEL DISPOSITIF

L'INVENTION EST RELATIVE A UN DISPOSITIF PERMETTANT LA RESTITUTION D'UN SON STEREOPHONIQUE OU PSEUDO-STEREOPHONIQUE A PLUSIEURS AUDITEURS4, 5, 6 SITUES DANS DES ZONES D'ECOUTE DIFFERENTES PAR RAPPORT A UN ENSEMBLE1 DE PRODUCTION DU SON. LE SON EST RESTITUE PAR UNE PAIRE D'ENCEINTES2, 3 ACOUSTIQUES. CHAQUE ENCEINTE COMPORTE DES HAUT-PARLEURS D'AIGUES ORIENTABLES SELECTIVEMENT DANS DES DIRECTIONS DIFFERENTES AUTOUR D'UN AXE VERTICAL ETOU HORIZONTAL EN DIRECTION DES ZONES OU SE TROUVENT LES DIVERS AUDITEURS. DE PLUS, CHAQUE HAUT-PARLEUR D'AIGUES COMPORTE DES MOYENS POUR VERIFIER L'ORIENTATION DE CELUI-CI.UNE GRILLE10. DANS LA GRILLE SONT MENAGES, A INTERVALLES REGULIERS, DES TROUS CIRCULAIRES12 DE RECEPTION DES MICROCOMPOSANTS. LE RECEPTACLE COMPORTE ENCORE DEUX PLAQUES DE FERMETURE14, 16 MONTEES CHACUNE DANS UNE DES ARMATURES ET DONT L'UNE AU MOINS EST DEPLACABLE PAR RAPPORT A L'ARMATURE CORRESPONDANTE, ET DES MOYENS ELASTIQUES24 POUR PRESSER LES PLAQUES DE FERMETURE CONTRE LA GRILLE, CHAQUE ...

DELAY ELEMENT

Process and device for the controlled distributing of sound between several sound sources

METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNAL

METHOD FOR REALIZING THREE DIMENSIONAL SOLID SOUND AND DEVICE THEREOF, ESPECIALLY REALIZING VIRTUAL SOLID SOUND AS TO MULTIPLE MOVING SOUND SOURCES

PURPOSE: A method for realizing three-dimensional solid sound and a device thereof are provided to realize the three-dimensional solid sound by assuring system stability and reducing calculation and complexity. CONSTITUTION: An ITD(Inter-aural Time Delay) module(10) gives inter-aural time delay(ITD) to more than one input sound signal. A weight adding module(20) multiplies output signals of the first step by principal component weight. A filtering module(30) filters the result of the second step by low order models of a number of basis vectors extracted from an HRTF(Head Related Transfer Function). © KIPO 2006 ...

3D sound Reproduction Apparatus and Method using Virtual Speaker Technique under Stereo Speaker Environments

APPARATUS AND METHOD FOR REPRODUCING SURROUND WAVE FIELD USING WAVE FIELD SYNTHESIS

Improving Apparatus and Method of Sound Quality for Portable Device

상이한 재생 라우드스피커 셋업에 대한 공간 오디오 신호의 세그먼트-와이즈 조정

... 본래의 라우드스피커 셋업에 대한 공간 오디오 신호(2)를, 본래의 라우드스피커 셋업과는 상이한 재생 라우드스피커 셋업으로 적응시키기 위한 장치(100)가 설명된다. 장치는, 본래의 라우드스피커 셋업의 세그먼트 내의 채널 신호들을 다이렉트 사운드(D) 및 앰비언스 컴포넌트들(A)로 분해하고, 다이렉트 사운드 컴포넌트들의 도달 방향을 결정하도록 구성된 다이렉트-앰비언스 분해기(130)를 포함한다. 다이렉트 사운드 렌더러(150)는, 재생 라우드스피커 셋업 정보를 수신하며, 재생 라우드스피커 셋업 내의 다이렉트 사운드 컴포넌트들의 인지된 도달 방향이 다이렉트 사운드 컴포넌트들의 도달 방향과 실질적으로 동일하도록 재생 라우드스피커 셋업 정보를 사용하여 다이렉트 사운드 컴포넌트들(D)을 조정한다. 결합기(180)는, 재생 라우드스피커 셋업의 라우드스피커들에 대한 라우드스피커 신호들을 획득하기 위해, 조정된 다이렉트 사운드 컴포넌트들 및 가급적 변경된 앰비언스 컴포넌트들을 결합한다.

AUDIO SYSTEM BASED ON THE LOCATION OF A USER AND A CONTROL METHOD THEREOF CAPABLE OF OUTPUTTING AN OPTIMIZED AUDIO SIGNAL FOR THE LOCATION OF A USER

PURPOSE: An audio system based on the location of a user and a control method thereof are provided to identify the location of a user by applying an RTLS(Real Time Location System) to a conventional audio system. CONSTITUTION: An audio system includes a plurality of speakers(12a,12b,12c,12d,12e,13a,13b) outputting sounds in a fixed position and a tag apparatus(16) which periodically transmits electric waves. The audio system includes a plurality of receivers(15a,15b,15c,15d,15e) receiving the electric waves from the tag apparatus and an audio processing part(11). The audio processing part calculates the location of the tag apparatus by analyzing the electric waves received from the receivers. The audio processing part outputs an audio signal to the plurality of speakers by adopting an equalizer effect corresponding to the location of the tag apparatus. COPYRIGHT KIPO 2012 ...

VIRTUAL AUDIO PROCESSING FOR LOUDSPEAKER OR HEADPHONE PLAYBACK

DECODING OF BINAURAL AUDIO SIGNALS

A method for synthesizing a binaural audio signal, the method comprising: inputting a parametrically encoded audio signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information describing a multi-channel sound image; and applying a predetermined set of head-related transfer function filters to the at least one combined signal in proportion determined by the corresponding set of side information to synthesize a binaural audio signal. A corresponding parametric audio decoder, parametric audio encoder, computer program product, and apparatus for synthesizing a binaural audio signal are also described. © KIPO & WIPO 2008 ...

SYSTEM AND METHOD FOR A 3D AUDIO IMPLEMENTATION OF A REAL TIME E-LEARNING SERVICE, CAPABLE OF SENSING A CURRENT SPEAKER FROM A PLURALITY OF REMOTE APPLICANTS EASILY

PURPOSE: A system and method for a 3D audio implementation of a real time e-learning service is provided to output different sounds at each speaker according to an image of the applicant displayed on the e-learning display. CONSTITUTION: A system and method for a 3D audio implementation of a real time e-learning service includes an e-learning display interface(100), an image detecting unit(200), and a sound gain controller(300). The e-learning display interface displays the image of an applicant attending the real time e-learning service. The image detecting unit extracts the position of the applicant image on the e-learning display interface. The sound gain controller controls the sound of the image at each speaker(400) according to extracted position. © KIPO 2009 ...

사운드 필드의 공간 성분들을 코딩하는 경우 사용하기 위한 코드북들의 식별

... 일반적으로, 사운드 필드의 공간 성분들을 압축하는 경우 사용될 코드북을 식별하기 위한 기법들이 설명된다. 하나 이상의 프로세서들을 포함하는 디바이스는 이 기법들을 수행하도록 구성될 수도 있다. 하나 이상의 프로세서들은, 복수의 공간 성분들 중 공간 성분을, 복수의 공간 성분들 중 나머지 성분들에 대한 공간 성분의 오더에 기초하여 압축하는 경우 사용하기 위해 허프만 코드북을 식별하도록 구성될 수도 있고, 이 공간 성분은 복수의 구면 조화 계수들에 대하여 벡터 기반 합성을 수행함으로써 생성된다.

CAMERA DRIVEN AUDIO SPATIALIZATION

AUDIO OUTPUTTING DEVICE, METHOD FOR CONTROLLING SAME, AND AUDIO OUTPUTTING SYSTEM

Disclosed are an audio outputting device, a method for controlling the same, and an audio outputting system. The method for controlling the audio outputting device for outputting multi-channel audio in association with at least one other audio outputting device according to the present invention, comprises the following steps: receiving profile information showing audio properties from at least one other audio outputting device; determining whether the received profile information is identical to pre-stored profile information of the audio outputting device; and reconfiguring the pre-stored profile information based on the received profile information according to a predetermined condition, when the received profile information is different from the pre-stored profile information. Accordingly, each of the audio outputting devices having different audio properties changes audio properties by considering the audio properties of a peripheral audio outputting device to adaptively form multiple ...

SPEAKER SOUND DISTRIBUTION APPARATUS AND METHOD

The present invention relates to a speaker sound distribution apparatus and method. The speaker sound distribution apparatus includes a setup signal transmission/reception part for transmitting a setup signal between a master speaker and a speaker sound distribution apparatus mounted on at least one or more slave speakers and receiving a response signal thereof, a temperature detection part for detecting a current ambient temperature, and a control part which calculates a distance and a direction of the slave speaker to the master speaker based on the response signal, recognizes a speaker arrangement space, sets a sound output time in consideration of a sound delay time corresponding to the detected temperature and speaker arrangement position, and transmits a sound signal according to the sound output time of each set speaker. It is possible to distribute a sound (or voice) signal for each speaker. COPYRIGHT KIPO 2017 (110) Setup signal transmission part (120) Setup signal receiving part ...

복수 단말의 멀티미디어 파일 협동 플레이 방법, 및 관련 장치 및 시스템

... 복수 단말의 멀티미디어 파일 협동 플레이 방법, 및 관련 장치 및 시스템이 개시된다. 멀티 단말 협동 플레이 시스템은 마스터 협동 플레이어, 플레이 그룹 관리 장치, 및 슬레이브 협동 플레이어를 포함한다. 플레이 그룹 관리 장치는 마스터 협동 플레이어에 의해 송신된 협동 플레이 요구에 따라 협동 플레이 식별자 및 플레이 위치 정보를 할당하고, 슬레이브 협동 플레이어에 의해 송신된 협동 플레이 참가 요구에 따라 플레이 위치 정보를 할당하며, 마스터 협동 플레이어는 플레이 그룹 관리 장치에 멀티미디어 파일 정보를 송신하며, 슬레이브 협동 플레이어는 플레이 그룹 관리 장치로부터 멀티미디어 파일 정보를 획득하며, 마스터 협동 플레이어는 슬레이브 협동 플레이에 플레이 커맨드를 송신하며, 이에 따라 마스터 협동 플레이어 및 슬레이브 협동 플레이어는 협동 플레이의 시작 시각에, 플레이 그룹 관리 장치에 의해 마스터 협동 플레이어 및 슬레이브 협동 플레이어에 할당된 각각의 플레이 위치에서, 멀티미디어 파일 정보에 대응하는 멀티미디어 파일을 플레이할 수 있다.

AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD

There is provided an audio encoding device capable of generating an appropriate monaural signal from a stereo signal while suppressing the lowering of encoding efficiency of the monaural signal. In a monaural signal generation unit (101) of this device, an inter-channel prediction/analysis unit (201) obtains a prediction parameter based on a delay difference and an amplitude ratio between a first channel audio signal and a second channel audio signal; an intermediate prediction parameter generation unit (202) obtains an intermediate parameter of the prediction parameter (called intermediate prediction parameter) so that the monaural signal generated finally is an intermediate signal of the first channel audio signal and the second channel audio signal; and a monaural signal calculation unit (203) calculates a monaural signal by using the intermediate prediction parameter. © KIPO & WIPO 2007 ...

DEVICE AND METHOD FOR EVALUATING AND OPTIMIZING SIGNALS ON THE BASIS OF ALGEBRAIC INVARIANTS

METHOD AND AN APPARATUS FOR ENCODING AND DECODING SURROUND EXPANSION DATA, PARTICULARLY FOR ENCODING OR DECODING AN AUDIO SIGNAL BY A SURROUND TYPE

PURPOSE: A method and an apparatus for encoding and decoding surround expansion data are provided to encode or decode additional information, capable of up-mixing an audio signal as a surround signal, as the expansion data. CONSTITUTION: An audio data decoding unit(705) decodes audio data hierarchically encoded. A termination code detecting unit(720) detects an identification code indicating that a payload of the audio data is terminated. A start code detecting unit(725) detects an identification code indicating that the payload of expansion data is started. An expansion type detecting unit(730) detects a type of the expansion data. A judging unit judges whether or not the detected type indicates additional information for decoding the audio data by a surround type. If the detected type indicates the additional information, an expansion data decoding unit(735) decodes expansion data corresponding to the additional information. © KIPO 2008 ...

자동차에서 3 차원 사운드 재생을 위한 라우드 스피커 배치

... 실시예는 차량용 라우드 스피커 재생 시스템을 제공한다. 라우드 스피커 재생 시스템은 제 1 라우드 스피커 어레이 및 제 2 라우드 스피커 어레이를 포함한다. 제 1 라우드 스피커 어레이는 제 1 스테레오 또는 가상 다중 채널 음장을 생성하도록 구성되며, 여기서 제 1 라우드 스피커 어레이는 차량의 운전자 위치를 향해 정렬된다. 제 2 라우드 스피커 어레이는 제 2 스테레오 또는 가상 다중 채널 음장을 생성하도록 구성되며, 여기서 제 2 라우드 스피커 어레이는 차량의 앞쪽 탑승자 위치를 향해 정렬된다.

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

descorrelação de sinal em um sistema de processamento de áudio

System for extraction of reverberant content of an audio signal

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

Device and method for improving stereophonic or pseudo-stereophonic audio signals

The proposed apparatus or the proposed method permits the linear variation of the degree of correlation particularly of pseudo-stereophonic audio signals, and overall provides a comprehensive, albeit extremely simple, postprocessing option. This is desirable in telephony, for example, which, even today, is still almost fundamentally based on a mono signal, in the area of professional postprocessing of audio signals, particularly for narrowing or expanding the mapping width thereof, for obtaining stable FM stereo signals, or else in the area of high quality electronic consumer goods, the aim of which is extremely simple but efficient handling.

Device and method for optimizing stereophonic or pseudo-stereophonic audio signals

The invention permits optimum choice of those parameters which form the basis for the generation of stereophonic or pseudo-stereophonic signals. The user is provided with means for stipulating the degree of correlation, the definition range, the loudness and also further parameters of the resulting signals according to psychoacoustic aspects, and hence for preventing artifacts. The invention can be used to define highly efficient encoders or decoders which confine audio signals intended for reproduction by two or more than two loudspeakers as a mono signal plus a few parameters. Specific areas of application are telecommunications (hands-free devices), global networks, computer systems, broadcasting and transmission devices, particularly satellite transmission devices, professional audio technology, television, film and broadcasting and also electronic consumer goods.

Method for Driving Loudspeakers

A circuit for operating loudspeakers includes a first, second, third and fourth loudspeaker circuit, having one input each for injecting a signal and one output each for connecting a loudspeaker input. The loudspeaker circuits are designed to amplify the injected signal and to provide the amplified signal at the outputs thereof. The loudspeaker circuits can, for example, be used for a 2.1 sound system. The three channels for a 2.1 sound system can be implemented by an amplifier circuit with four loudspeaker circuits, one loudspeaker circuit each being required for the two stereo channels left and right. A subwoofer channel can be driven differentially by two loudspeaker circuits. The stereo channels are, by contrast, only still connected to one loudspeaker circuit each, and so the stereo channels require at least one further common ground cable.

Speaker Apparatus

A speaker apparatus includes speaker units, a first inputting section to which one channel audio signal is supplied, a second inputting section to which at least L-channel and R-channel audio signals are supplied, an acoustic effect imparting section that performs a first signal processing on the audio signal supplied to the first inputting section to output L channel and R channel audio signals, an outputting section that outputs the audio signal supplied to the first inputting section to at least one of the speaker units, and a widening effect imparting section that performs a second signal processing for imparting a sound image widening effect on both the L-channel and R-channel audio signals and the at least L-channel and R-channel audio signals and supplies processed audio signals to speaker units for corresponding channels respectively.

Camera Driven Audio Spatialization

A camera input can be used by the computer to support audio spatialization or to improve audio spatialization of an application that already supports it. A computer system may to support audio spatialization, for example, by modifying the relative latency or relative amplitude of the rendered audio packets. If a sound is intended, for example, to be located on the left side of the user, then the audio channel that is rendered on the headset speaker located on the user's left ear may have a somewhat decreased latency and increased amplitude compared to the other audio channel.

APPARATUS AND METHOD FOR GENERATING PANORAMIC SOUND

An apparatus and method for generating a panoramic sound are provided. The panoramic sound generation apparatus may include a panning coefficient calculation unit to calculate a panning coefficient that represents directivity of a sound source using an input signal, a masker determination unit to determine a direction masker that extracts a sound source of a desired direction based on the panning coefficient, and a channel separation unit to separate to be used as an output signal, output to a sound output device, using the direction masker. 1. A panoramic sound generation apparatus comprising:a panning coefficient calculation unit, using at least one processor, to calculate a panning coefficient that represents directivity of a sound source using an input signal;a masker determination unit to determine a direction masker that extracts a sound source of a desired direction based on the panning coefficient; anda channel separation unit to separate the input signal to be used as an output signal, and to output the output signal to a sound output device, using the direction masker.2. The panoramic sound generation apparatus of claim 1 , wherein the panning coefficient calculation unit calculates the panning coefficient using a frequency component based on the input signal.3. The panoramic sound generation apparatus of claim 1 , wherein the masker determination unit comprises:a control coefficient determiner to determine a control coefficient for windowing the panning coefficient using configuration information of the sound output device; anda window processor to process a panning coefficient window based on the sound output device using the control coefficient.4. The panoramic sound generation apparatus of claim 3 , wherein the control coefficient determiner determines the control coefficient using a first angle based on a first sound output device corresponding to a left boundary and a second sound output device corresponding to a right boundary claim 3 , a second ...

Downmix Limiting

The invention relates to downmixing techniques by which output audio signals are obtained from input audio signals partitioned into subgroups. A variable common gain limiting factor is applied to all downmix coefficients that govern the contributions from the input signals in a subgroup. While preserving the proportions between signal values within a subgroup, the invention makes it possible to limit the gain of different input signal subgroups to different extents, so that relatively more perceptible signals can be limited relatively less. It then becomes possible to achieve a consistent dialogue level while transitioning in a less perceptible fashion between signal portions with and without gain limiting. Embodiments of the invention include a method, a mixing system and a computer-program product.

AUDIO SIGNAL PROCESSING DEVICE

A postprocessor mixes audio signals for an LSch and an RSch indicating virtual surround sounds generated by a virtual surround processing unit with audio signals of an Rch, an Lch, and a Cch, performs a high-pass filter process, and then outputs the processed audio signals as audio signals for an Rsp and an Lsp which are main speakers. Further, the postprocessor performs phase inversion on the audio signals for the LSch and the RSch indicating the virtual surround sounds generated by the virtual surround processing unit, mixes the processed audio signals with audio signals of the Rch, the Lch, the Cch, and an LFEch, performs a low-pass filter process on the mixed signals, and then outputs the processed audio signals as audio signals for an SW which is a subwoofer. Thus, in a sound field realized by a virtual surround system, an influence of the sounds emitted from the subwoofer on localization of sounds of channels of virtual speakers is reduced. 1. An audio signal processing device comprising:a generation unit configured to generate n (where n is a natural number equal to or greater than 2) virtual process audio signals by processing at least one input audio signal so as to localize the input audio signal at a virtual position; andan output unit configured to output the n virtual process audio signals generated by the generation unit to n speakers as first output signals and to output the n virtual process audio signals generated by the generation unit to a subwoofer as second output signals,wherein the output unit is configured to output the first output signals and the second output signals so as to have mutually opposite phases.2. The audio signal processing device according to claim 1 , wherein the output unit includes a phase inversion processing unit configured to invert the phase of the first output signals or the second output signals so that the first output signals and the second output signals have opposite phases to each other.3. The audio signal ...

AUTOMATIC ANNOUNCER VOICE ATTENUATION IN A PRESENTATION OF A BROADCAST EVENT

A method of automatic announcer voice removal from a televised sporting event. A sound processing circuit divides an audio input signal of a televised sporting event into multiple audio segments. The audio input signal includes crowd noise and announcer commentary. If an audio segment does not exceed a pre-defined amplitude threshold, a voice removal utility adds the audio segment to a recent crowd noise library and stores the segment in an output buffer. If the amplitude of a segment exceeds the threshold, the utility adds the segment to a recent announcer voice library. The sound processing circuit generates an attenuated version of the segment and blends the attenuated version with one or more mixed segments from the recent crowd noise library. The voice removal utility stores the attenuated and blended segment in the output buffer and outputs one or more audio segments from the buffer in a chronological order. 1. In a computer system , a method comprising:dividing an audio input signal corresponding to a broadcast event into a temporally sequential plurality of audio segments, wherein at least some of the plurality of audio segments include both crowd noise and announcer commentary, and wherein said announcer commentary has a greater amplitude than said crowd noise in multiple of the plurality of audio segments; adding said audio segment to a recent crowd noise library; and', 'storing said audio segment in an output buffer;, 'in response to a determination that an amplitude of an audio segment from the plurality of audio segments does not exceed a pre-defined amplitude threshold adding said audio segment to a recent announcer voice library;', 'generating an attenuated version of said audio segment;', 'blending said attenuated version of said audio segment with one or more mixed audio segments included in said recent crowd noise library; and', 'storing said attenuated version of said audio segment blended with said one or more mixed audio segments in said output ...

Diffusing Acoustical Crosstalk

When two loudspeakers play the same signal, a “phantom center” image is produced between the speakers. However, this image differs from one produced by a real center speaker. In particular, acoustical crosstalk produces a comb-filtering effect, with cancellations that may be in the frequency range needed for the intelligibility of speech. Methods for using phase decorrelation to fill in these gaps and produce a flatter magnitude response are described, reducing coloration and potentially enhancing dialogue clarity. These methods also improve headphone compatibility and reduce the tendency of the phantom image to move toward the nearest speaker. 1. A method of decorrelating a signal using phase diffusion at high frequencies , the method comprising:separating a left input signal into a left high-frequency signal and a left low-frequency signal and separating a right input signal into a right high-frequency signal and a right low-frequency signal;creating a diffused left high-frequency signal from the left high-frequency signal using a first diffuser;creating a diffused right high-frequency signal from the right high-frequency signal using a second diffuser;creating a delayed left low-frequency signal and a delayed right low-frequency signal wherein a delay of the delayed left low-frequency signal and a delay of the delayed right low-frequency signal are related to a delay of the first diffuser and a delay of the second diffuser; andcombining the delayed left low-frequency signal with the diffused left high-frequency signal and combining the delayed right low-frequency signal with the diffused right high-frequency signal, thereby producing a stereo signal with phase diffusion at high frequencies.2. The method of further comprising accepting a left input signal and a right input signal.3. The method of wherein the first diffuser includes a first allpass filter and the second diffuser includes a second allpass filter.4. The method of further comprising:applying in one of ...

AUDIO SIGNAL REPRODUCTION DEVICE AND AUDIO SIGNAL REPRODUCTION METHOD

An audio signal reproduction device generates, from an obtained audio signal, first reproduction signals for the first speaker group, sounds from which are localized at first virtual sound positions, and second reproduction signals for the second speaker group, sounds from which are localized at second virtual sound positions substantially the same as the first virtual sound positions. The audio signal reproduction device generates the first reproduction signals and the second reproduction signals so that at least phases or sound pressure values of a first sound and a second sound are different at a listening position, the first sound being indicated by the first reproduction signals, and localized at a first position among the first virtual sound positions, the second sound being indicated by the second reproduction signals, localized at a substantially same position as the first position, and substantially the same as the first sound. 1. An audio signal reproduction device which reproduces an audio signal using a first speaker group which includes plural speakers placed in vicinity of a listener and a second speaker group which includes plural speakers and is placed closer to the listener than the first speaker group is , the audio signal including position information indicating , for plural audio channels , positions of virtual sounds to be localized , the audio signal reproduction device comprising:an obtaining unit configured to obtain the audio signal; anda virtual sound field generation unit configured to generate, by performing signal processing on the audio signal, first reproduction signals for the first speaker group, sounds from which are localized at first virtual sound positions, and second reproduction signals for the second speaker group, sounds from which are localized at second virtual sound positions substantially the same as the first virtual sound positions,wherein the virtual sound field generation unit is configured to generate the first ...

SPATIALIZATION USING STEREO DECORRELATION

A method for spatialization using stereo decorrelation. The method generates decorrelated left and right signals by filtering the left input channel using an all-pass filter with a first group delay and filtering the right input channel using an all-pass filter with a second group delay; the second group delay being different than that of the first group delay. The decorrelated left signal is subtracted from the decorrelated right signal to create a second left signal and the decorrelated right signal is subtracted from the decorrelated left signal to create a second right signal. The method generates a side signal by adding the filtered left input channel signal and the filtered right input channel signal. The side signal is added to the second left signal to generate a left output signal the side signal is added to the second right signal to generate a right output signal. 1. A method for spatialization using stereo decorrelation , in an audio system having left and right input channels , the method comprising: filtering the left input channel using an all-pass filter with a first group delay;', 'filtering the right input channel using an all-pass filter with a second group delay; the second group delay being different than that of the first group delay;, 'generating decorrelated left and right signals bysubtracting the decorrelated left signal from the decorrelated right signal to create a second left signal;subtracting the decorrelated right signal from the decorrelated left signal to create a second right signal;generating a side signal by adding the filtered left input channel signal and the filtered right input channel signal;adding the side signal to the second left signal to generate a left output signal; and,adding the side signal to the second right signal to generate a right output signal.2. The method of further comprising applying gain to the second left and right signals claim 1 , and the side signal.3. The method of where the gain applied to the side ...

SYSTEM AND METHOD FOR REPRODUCING WAVE FIELD USING SOUND BAR

Disclosed is a system and method for reproducing a wave field that reproduces a wave field using a sound bar, the system including an input signal analyzer to divide an input signal into an audio signal for a plurality of channels, and identify a position of a loud speaker corresponding to the plurality of channels, a rendering unit to process the audio signal for the plurality of channels, using a rendering algorithm based on the position, and generate an output signal, and a loud speaker array to output the output signal using loud speakers corresponding to the plurality of channels, and reproduce a wave field. 1. A system for reproducing a wave field , the system comprising:an input signal analyzer to divide an input signal into an audio signal for a plurality of channels, and identify a position of a loud speaker corresponding to the plurality of channels;a rendering unit to process the audio signal for the plurality of channels, using a rendering algorithm based on the position, and generate an output signal; anda loud speaker array to output the output signal via the loud speaker corresponding to the plurality of channels, and reproduce a wave field.2. The system of claim 1 , wherein the rendering unit processes an audio signal for a plurality of channels corresponding to a forward channel claim 1 , using a wave field synthesis rendering algorithm claim 1 , and processes an audio signal for a plurality of channels corresponding to a side channel or a rear channel.3. The system of claim 2 , wherein the rendering unit determines a position at which a focused sound source is to be generated based on a listening space in which a wave field is reproduced when an audio signal for a plurality of channels is processed claim 2 , using a focused sound source rendering algorithm.4. The system of claim 1 , wherein the rendering unit selects from among a focused sound source rendering algorithm claim 1 , a beam-forming rendering algorithm claim 1 , and a deccorelator ...

Angle-dependent operating device or method for generating a pseudo-stereophonic audio signal

An apparatus for stereophonizing a mono signal, comprising a convener for converting the mono signal into a principal signal and a side signal by delaying and amplifying the mono signal on the basis of an angle φ between sound source and microphone principal axis. 1. An apparatus for stereophonizing a mono signal , comprisinga converter for converting the mono signal into a principal signal and a side signal by delaying and amplifying the mono signal on the basis of an angle φ between sound source and microphone principal axis.2. Apparatus according to claim 1 , wherein the converter is configured to convert the mono signal into a principal signal and a side signal by delaying and amplifying the mono signal on the basis of an opening angle α claim 1 , which adjoins the microphone principal axis on one side.3. Apparatus according to claim 2 , wherein the convener is configured to convert the mono signal into a principal signal and a side signal by delaying and amplifying the mono signal on the basis of an further opening angle β claim 2 , which adjoins the microphone principal axis on another side.4. Apparatus according to claim 3 , wherein the opening angle α and the further opening angle βare equal.5. Apparatus according to claim 1 , wherein the converter is configured to convert the mono signal into a principal signal and a side signal by delaying and amplifying the mono signal on the basis of a directivity pattern of the mono signal.6. Apparatus according to claim 1 , wherein the converter is configured to convert the mono signal into a principal signal and a side signal by delaying and amplifying the mono signal on the basis of an opening angle α claim 1 , which adjoins the microphone principal axis on the left claim 1 , an further opening angle β claim 1 , which adjoins the microphone principal axis on the right and a directivity pattern of the mono signal.7. Apparatus according to claim 6 , wherein the angle α is zero claim 6 , and the opening angle α and the ...

NAVIGATION WITH THREE DIMENSIONAL AUDIO EFFECTS

Mechanisms for navigation via three dimensional audio effects are described. A current location of a device and a first point of interest are determined. The point of interest may be determined based on a web service and the current location of the device may be determined via mobile device signals. A zone that includes the point of interest may be determined. A three dimensional audio effect that simulates a sound being emitted from the zone may be generated. The three dimensional audio effect may be transmitted to speakers capable of simulating three dimensional audio effects. The transmitted three dimensional audio effect may aid in navigation from a current location to the point of interest. 1. A computer-implemented method comprising:determining a current location of a device;determining a location of a first point of interest;determining a zone, the zone including the location of the first point of interest;generating a three dimensional audio effect, the three dimensional audio effect to represent a sound emitted from a point in the zone;sending the three dimensional audio effect.2. The computer-implemented method of claim 1 , wherein: sending a web service request with an identifier associated with the first point of interest; and', 'receiving a location associated with the first point of interest., 'determining a location of the first point of interest comprises3. The computer-implemented method of claim 1 , wherein:determining a current location of the device comprises analyzing mobile device signals.4. The computer-implemented method of claim 1 , wherein:sending the three dimensional audio effect further comprises sending a web service message, the web service message including the three dimensional audio effect.5. The computer-implemented method of claim 1 , further comprising:determining a distance between the current location of the device and the first point of interest;wherein the three dimensional audio effect further simulates sound emitted from ...

Methods for Audio Signal Transient Detection and Decorrelation Control

Some audio processing methods may involve receiving audio data corresponding to a plurality of audio channels and determining audio characteristics of the audio data, which may include transient information. An amount of decorrelation for the audio data may be based, at least in part, on the audio characteristics. If a definite transient event is determined, a decorrelation process may be temporarily halted or slowed. Determining transient information may involve evaluating the likelihood and/or the severity of a transient event. In some implementations, determining transient information may involve evaluating a temporal power variation in the audio data. Explicit transient information may or may not be received with the audio data, depending on the implementation. Explicit transient information may include a transient control value corresponding to a definite transient event, a definite non-transient event or an intermediate transient control value.

REDUCING CORRELATION BETWEEN HIGHER ORDER AMBISONIC (HOA) BACKGROUND CHANNELS

In general, techniques are described for compression and decoding of audio data are generally disclosed. An example device for compressing audio data includes one or more processors configured to apply a decorrelation transform to ambient ambisonic coefficients to obtain a decorrelated representation of the ambient ambisonic coefficients, the ambient HOA coefficients having been extracted from a plurality of higher order ambisonic coefficients and representative of a background component of a soundfield described by the plurality of higher order ambisonic coefficients, wherein at least one of the plurality of higher order ambisonic coefficients is associated with a spherical basis function having an order greater than one. 1. A method comprising:obtaining a decorrelated representation of ambient ambisonic coefficients having at least a left signal and a right signal, the ambient ambisonic coefficients having been extracted from a plurality of higher order ambisonic coefficients and representative of a background component of a soundfield described by the plurality of higher order ambisonic coefficients, wherein at least one of the plurality of higher order ambisonic coefficients is associated with a spherical basis function having an order greater than one; andgenerating a speaker feed based on the decorrelated representation of the ambient ambisonic coefficients.2. The method of claim 1 , further comprising applying a recorrelation transform to the decorrelated representation of the ambient ambisonic coefficients to obtain a plurality of correlated ambient ambisonic coefficients.3. The method of claim 2 , wherein applying the recorrelation transform comprises applying an inverse phase-based transform to the ambient ambisonic coefficients.4. The method of claim 3 , wherein the inverse phase-based transform has been normalized according to one of N3D (full three-D) normalization.5. The method of claim 3 , wherein the inverse phase-based transform has been normalized ...

RENDERING OF AUDIO OBJECTS WITH APPARENT SIZE TO ARBITRARY LOUDSPEAKER LAYOUTS

Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment. 146-. (canceled)47. A method , comprising:receiving audio reproduction data comprising one or more audio objects, the audio objects comprising audio signals and associated metadata, the metadata including at least audio object position data and audio object size data;computing, for an audio object from the one or more audio objects, virtual source gain values from virtual sources at respective virtual source locations within an audio object area or volume defined by the audio object position data and the audio object size data; andcomputing a set of audio object gain values for each of a plurality of output channels based, at least in part, on the computed virtual source gain values, wherein each output channel corresponds to at least one reproduction speaker of a reproduction environment and each of said virtual source locations corresponds to a respective static location within the reproduction environment.48. The method of claim 47 , wherein the process of computing the set of audio object gain values involves computing a ...

AUDIO PROVIDING APPARATUS AND AUDIO PROVIDING METHOD

An audio providing apparatus and method are provided. The audio providing apparatus includes: an object renderer configured to render an object audio signal based on geometric information regarding the object audio signal; a channel renderer configured to render an audio signal having a first channel number into an audio signal having a second channel number; and a mixer configured to mix the rendered object audio signal with the audio signal having the second channel number. 1. An audio providing method comprising:receiving a plurality of input channel signals;aligning a difference in phase between correlated input channel signals among the plurality of input channel signals; andconverting the plurality of input channel signals including the correlated input channel signals into a plurality of output channel signals based on an input layout and an output layout,wherein the input layout is a format of the plurality of input channel signals and the output layout is a format of the plurality of output channel signals.2. The method of claim 1 , wherein the output layout is 2D layout.3. The method of claim 1 , wherein the plurality of output channel signals include a virtual output channel signal to reproduce a height input channel signal.4. The method of claim 1 , wherein the plurality of input channel signals comprise information for determining whether to perform virtual 3D rendering on a specific frame. This is a continuation of U.S. application Ser. No. 14/649,824 filed on Jun. 4, 2015, which is a National Stage application under 35 U.S.C. §371 of PCT/KR2013/011182, filed on Dec. 4, 2013, which claims the benefit of U.S. Provisional Application No. 61/732,938, filed on Dec. 4, 2012 in the United States Patent and Trademark Office, and U.S. Provisional Application No. 61/732,939, filed on Dec. 4, 2012 in the United States Patent and Trademark Office, all the disclosures of which are incorporated herein in their entireties by reference.Apparatuses and methods ...

SIMULATING ACOUSTIC OUTPUT AT A LOCATION CORRESPONDING TO SOURCE POSITION DATA

Systems and methods of simulating acoustic output at a location corresponding to source position data are disclosed. A particular method includes receiving an audio signal and source position data associated with the audio signal. A set of speaker signals are applied to a plurality of speakers, where the set of speaker driver signals causes the plurality of speakers to generate acoustic output that simulates output of the audio signal by an audio source at a location corresponding to the source position data. 1. A method comprising:receiving an audio signal and source position data associated with the audio signal; andapplying a set of speaker driver signals to a plurality of speakers, wherein the set of speaker driver signals causes the plurality of speakers to generate acoustic output that simulates output of the audio signal by an audio source at a location corresponding to the source position data.2. The method of claim 1 , wherein the set of speaker driver signals corresponds to one or more fixed speakers claim 1 , one or more virtual speakers claim 1 , or a combination thereof3. The method of claim 1 , wherein the location corresponding to the source position data is distinct from locations of the plurality of speakers.4. The method of claim 1 , further comprising applying a second set of speaker driver signals to the plurality of speakers to generate acoustic output corresponding to a second location that is different from the location.5. The method of claim 1 , wherein the audio signal claim 1 , the source position data claim 1 , or both are received from an automatic driver assistance system claim 1 , a navigation system claim 1 , or a mobile device.6. The method of claim 1 , further comprising:up-mixing the audio signal to generate a plurality of intermediate signal components;down-mixing the plurality of intermediate signal components to generate a plurality of speaker signal components; andprocessing the plurality of speaker signals components to ...

Simulating Acoustic Output at a Location Corresponding to Source Position Data

An audio system includes a plurality of speakers and an audio signal processor. The audio signal processor receives an audio signal and source position data associated with the audio signal; and applies a set of speaker driver signals to the plurality of speakers. The set of speaker driver signals causes the plurality of speakers to generate acoustic output that simulates output of the audio signal by an audio source at a location corresponding to the source position data in each of a plurality of discrete acoustic areas. 1. An audio system comprising:a plurality of speakers; and receive an audio signal and source position data associated with the audio signal; and', 'apply a set of speaker driver signals to the plurality of speakers, wherein the set of speaker driver signals causes the plurality of speakers to generate acoustic output that simulates output of the audio signal by an audio source at a location corresponding to the source position data in each of a plurality of discrete acoustic areas., 'an audio signal processor configured to2. The audio system of claim 1 , wherein the audio signal processor is configured to receive user input indicating engagement of a privacy mode claim 1 , and claim 1 , in response claim 1 , halt the simulation of output of the audio signal in a first one of the acoustic areas.3. The audio system of claim 2 , wherein claim 2 , in response to receiving the user input indicating engagement of the privacy mode claim 2 , the audio signal processor is configured to override the received source position data with predetermined position data claim 2 , such that output of the audio signal is simulated at a location corresponding to the predetermined position data in a second one of the acoustic areas.4. The audio system of claim 3 , wherein the predetermined position data corresponds to a location in the second one of the acoustic areas in which crosstalk is substantially attenuated relative to the first one of the acoustic areas.5. The ...

DECODING OF AUDIO SCENES

Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder. 1. A method for decoding an audio scene represented by N audio signals , the method comprising:receiving a bit stream comprising L auxiliary signals, M downmix signals and matrix elements of a reconstruction matrix, wherein the matrix elements are transmitted as side information in the bit stream, and wherein at least some of the M downmix signals are formed from two or more of the N audio signals;generating the reconstruction matrix using the matrix elements; and 'wherein M is less than N, and M is equal or greater than one.', 'reconstructing the N audio signals from the M downmix signals and the L auxiliary signals using the reconstruction matrix, wherein approximations of the N audio signals are obtained as linear combinations of the M downmix signals with the matrix elements of the reconstruction matrix as coefficients in the linear combinations,'}2. The method of claim 1 , wherein at least some of the N audio signals are rendered to generate a three-dimensional audio environment.3. The method of claim 1 , wherein the audio scene comprises a three-dimensional audio environment which includes audio elements being associated with positions in a three-dimensional space that can be rendered for playback on an audio system.4. The method of claim 1 , wherein the M downmix signals are arranged in a first portion of a bit stream using a first format and the matrix elements are arranged in a second field of the bit stream using a second format.5. The method of claim 1 , wherein the linear combinations are formed by multiplying a matrix of the M ...

Apparatus and method for processing stereo signals for reproduction in cars to achieve individual three-dimensional sound by frontal loudspeakers

Embodiments provide a digital processor including an ambient portion extractor and a spatial effect processing stage. The ambient portion extractor is configured to extract an ambient portion from a multi-channel signal. The spatial effect processing stage is configured to generate a spatial effect signal based on the ambient portion of the multi-channel signal. The digital processor is configured to combine the multi-channel signal or a processed version thereof with the spatial effect signal.

Dynamic Audio Upmixer Parameters For Simulating Natural Spatial Variations

A system and method for creating natural spatial variations in an audio output. At least one parameter in a set of mixer tuning parameters is dynamically modified over time and within a predetermined range that is defined by a set of modification control parameters. The set of mixer tuning parameters that includes the at least one dynamically modified parameter is applied to a mixer allowing the mixer to create natural spatial variations in the audio output to be played at one or more loudspeakers. 1. A system for creating natural spatial variations in an audio output , comprising:an audio input signal; andan audio signal processor that receives the audio input signal, the audio signal processor is configured lo dynamically modify at least one parameter in a set of mixer tuning parameters over time and within a predetermined range to transform the audio input signal into an audio output having natural spatial variations in the audio output.2. The system of wherein the at least one parameter is modified in real time.3. The system of wherein the at least one parameter in the set of mixer tuning parameters is modified over time and within a predetermined range by a dynamic parameter modification algorithm that also applies a set of modification control parameters.4. The system of wherein the dynamic parameter modification algorithm communicates the at least one dynamically modified parameter to a mixer.5. The system of wherein the mixer is a surround upmixer.6. The system of further comprises:a conventional parameter management algorithm that communicates the set of mixer tuning parameters and the modification control parameters to the audio signal processor; andthe dynamic parameter modification algorithm is carried out in the audio signal processor directly by the mixer.7. The system of wherein the predetermined range is based on at least one variable that is external to the set of mixer tuning parameters claim 1 , the predetermined range may be modified based on a ...

SYSTEM AND METHOD FOR EVALUATING MEDIA SEGMENTS FOR INTERESTINGNESS

Disclosed herein are systems, methods, and computer-readable storage device for analyzing a first audiofile, to yield a first analysis, wherein the first analysis identifies a first segment of a first plurality of segments within the first audiofile, the first segment being one of a most interesting segment, a most important segment, a most relevant segment, and a most representative segment. A same analysis can be performed on a second audiofile, to yield a second analysis, wherein the second analysis identifies a second segment of a second plurality of segments within the second audiofile, the second segment comprising one of a most interesting segment, a most important segment, a most relevant segment, and a most representative segment. The first segment is presented as a representative segment of the first audiofile and the second segment is presented as a representative of the second audiofile within a three-dimensional audio space. 1. A method comprising:analyzing a first audiofile, to yield a first analysis, wherein the first analysis identifies a first segment of a first plurality of segments within the first audiofile, the first segment being one of a most interesting segment, a most important segment, a most relevant segment, and a most representative segment;analyzing a second audiofile, to yield a second analysis, wherein the second analysis identifies a second segment of a second plurality of segments within the second audiofile, the second segment comprising one of a most interesting segment, a most important segment, a most relevant segment, and a most representative segment; andpresenting the first segment as a representative segment of the first audiofile and the second segment as a representative of the second audiofile within a three-dimensional audio space.2. The method of claim 1 , further comprising:presenting the first segment and the second segment within the three-dimensional audio space simultaneously, wherein the presenting comprises ...

DECODING OF AUDIO SCENES

Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder. 1. A method for decoding an audio scene represented by N audio signals , the method comprising:receiving a bit stream comprising L auxiliary signals, M downmix signals and matrix elements of a reconstruction matrix, wherein the matrix elements are transmitted as side information in the bit stream;generating the reconstruction matrix using the matrix elements; andreconstructing the N audio signals from the M downmix signals and the L auxiliary signals using the reconstruction matrix, wherein approximations of the N audio signals are obtained as linear combinations of the M downmix signals with the matrix elements of the reconstruction matrix as coefficients in the linear combinations,wherein M is less than N, and M is equal or greater than one.2. The method of wherein at least some of the M downmix signals are formed from two or more of the N audio signals.3. The method of wherein at least some of the N audio signals are rendered to generate a three-dimensional audio environment.4. The method of wherein the audio scene comprises a three-dimensional audio environment which includes audio elements being associated with positions in a three-dimensional space that can be rendered for playback on an audio system.5. The method of wherein the M downmix signals are arranged in a first portion of a bit stream using a first format and the matrix elements are arranged in a second field of the bit stream using a second format.6. The method of wherein the linear combinations are formed by multiplying a matrix of the M downmix signals with the reconstruction ...

METHOD FOR PROCESSING OF SOUND SIGNALS

A method for processing audio signals for creating a three dimensional sound environment includes: receiving at least one input signal from at least one sound source; creating a simulated signal at least in part on the basis of the received at least one input signal, the simulated signal representing a simulation of at least one input signal reflecting from the ground or floor; and creating an output signal at least partly on the basis of the simulated signal and the at least one received input signal, the output signal including a plurality of audio channels; at least two channels of the audio channels of the output signal representing signals for sound transducers above a listener's ear level at a nominal listening position, and at least two channels of the audio channels of the output signal representing signals for sound transducers below a listeners ear level at a nominal listening position. 1. A method for processing audio signals for creating a three dimensional sound environment , comprising at least the steps ofreceiving at least one input signal from at least one sound source,creating a simulated signal at least in part on the basis of said received at least one input signal, said simulated signal representing a simulation of at least one input signal reflecting from the ground or a floor, andcreating an output signal at least partly on the basis of said simulated signal and said at least one received input signal, said output signal comprising a plurality of audio channels;at least two channels of said audio channels of said output signal representing signals for sound transducers above a listener's ear level at a nominal listening position, andat least two channels of said audio channels of said output signal representing signals for sound transducers below a listener's ear level at a nominal listening position.2. A method according to claim 1 , further comprising at least the steps ofcreating output signals for a background sound environment byreceiving ...

REAL-TIME IMMERSIVE MEDIATED REALITY EXPERIENCES

The invention relates to creating real-time, immersive mediated reality environments using real data collected from a physical event or venue. The invention provides a virtual participant with the ability to control their viewpoint and freely explore the venue, in real time by synthesizing virtual data corresponding to a requested virtual viewpoint using real images obtained from data collectors or sources at the venue. By tracking and correlating real and virtual viewpoints of virtual participants, physical objects, and data sources, systems and methods of the invention can create photo-realistic images for perspective views for which there is not physically present data source. Systems and methods of the invention also relate to applying effect objects to enhance the immersive experience including virtual guides, docents, text or audio information, expressive auras, tracking effects, and audio. 1. A system for creating a mediated reality environment , said system comprising a server computing system comprising a processor coupled to a tangible , non-transitory memory , the system operable to:receive, in real-time, real viewpoint information for one or more data collectors located at a venue;receive a virtual viewpoint from a computing device of a virtual participant, said computing device comprising a processor coupled to a tangible, non-transitory memory;receive one or more real-time images from the one or more data collectors where the one or more data collectors have a real viewpoint which intersects the virtual viewpoint, said one or more real-time images comprising a plurality of real pixels;create, using the server's processor, a real-time virtual image comprising a plurality of virtual pixels and corresponding to the virtual viewpoint by using pixel information from the one or more real-time images; andcause the computing device of the virtual participant to display the real-time virtual image.2. The system of claim 1 , further operable to:identify, using ...

METHOD FOR AUDIO REPRODUCTION IN A MULTI-CHANNEL SOUND SYSTEM

The invention relates to a method for reproducing audio in a multi-channel sound system including two input signals (L and R), wherein output signals are generated for different sound perception levels. In order to develop said method in such a way that audio can be reproduced within a larger range of applications in a multi-channel sound system, according to the invention, only a lower sound perception level () and a higher sound perception level () are generated, and a maximum of six output signals are generated, a maximum of two output signals being allocated to the lower sound perception level () and a maximum of four output signals being allocated to the higher sound perception level (). 17676. A method for audio reproduction in a multi-channel sound system comprising two input signals L and R , wherein output signals are generated for different listening levels , characterized in that only one lower listening level () and only one upper listening level () are generated , wherein a maximum of six output signals , with a maximum of two output signals for the lower listening level () and a maximum of four output signals for the upper listening level () , are generated.276. The method according to claim 1 , characterized in that stereo signals are generated for the signals in the lower listening level () and upper listening levels ().376. The method according to claim 1 , characterized in that mono signals are generated for the signals in the lower listening level () and upper listening level ().47. The method according to claim 1 , characterized in that mono signals are generated for the signals in the lower listening level ().57. The method according to claim 1 , characterized in that mono signals are generated for the signals in the upper listening level ().6. The method according to one of the to claim 1 , characterized in that the output signals serve as further input signals.7. The method according to one of the to claim 1 , characterized in that claim 1 , ...

APPARATUS AND METHOD FOR PROCESSING MULTI-CHANNEL AUDIO SIGNAL

Disclosed is an apparatus and method for processing a multichannel audio signal. A multichannel audio signal processing method may include: generating an N-channel audio signal of N channels by down-mixing an M-channel audio signal of M channels; and generating a stereo audio signal by performing binaural rendering of the N-channel audio signal. 1. A multichannel audio signal processing method , comprising:generating an N-channel audio signal of N channels by down-mixing an M-channel audio signal of M channels; andgenerating a stereo audio signal by performing binaural rendering of the N-channel audio signal.2. The method of claim 1 , wherein the generating of the stereo audio signal comprises:generating channel-by-channel stereo audio signals using filters corresponding to playback locations of channel-by-channel audio signals of the N channels; andgenerating the stereo audio signal by mixing the channel-by-channel stereo audio signals.3. The method of claim 1 , wherein the generating of the stereo audio signal comprises generating the stereo audio signal using a plurality of binaural renderers respectively corresponding to the channels of the N-channel audio signal.4. A multichannel audio signal processing method claim 1 , comprising:sub-sampling the number of channels of a multichannel audio signal based on a virtual loudspeaker layout; andgenerating a stereo audio signal by performing binaural rendering of the sub-sampled multichannel audio signal.5. The method of claim 4 , wherein the generating of the stereo audio signal comprises performing binaural rendering of the sub-sampled multichannel audio signal in a frequency domain.6. The method of claim 4 , wherein the generating of the stereo audio signal comprises generating the stereo audio signal using a plurality of binaural renderers respectively corresponding to the channels of the N-channel audio signal.7. A multichannel audio signal processing method claim 4 , comprising:sub-sampling the number of channels ...

AUDIO PROCESSING APPARATUS

An example approach for navigation is provided with an apparatus comprising: a navigation processor configured to generate at least one navigation command; and an audio processor configured to determine at least one directional audio parameter so as to provide a direction, the at least one directional audio parameter being based on the navigation command and further configured to process at least one audio signal based on the at least one directional audio parameter such that a virtual sound image of a navigation routing is generated based on the processed at least one audio signal. 1. A method comprising:generating at least one navigation command;determining at least one directional audio parameter so as to provide a direction, the at least one directional audio parameter being based on the navigation command; andprocessing at least one audio signal based on the at least one directional audio parameter such that a virtual sound image of a navigation routing is generated based on the processed at least one audio signal.2. The method as claimed in claim 1 , wherein the at least one directional audio parameter comprises at least one of:at least one interaural level difference based on the at least one navigation command;at least one interaural time difference based on the at least one navigation command;at least one interaural phase difference based on the at least one navigation command; anda head related transfer function based on the at least one navigation command.3. The method as claimed in claim 1 , wherein the at least one navigation command further comprises at least one directional indication associated with the at least one audio signal claim 1 , and wherein determining at least one directional audio parameter is based on the at least one directional indication.4. The method as claimed in claim 1 , wherein the at least one navigation command further comprises at least one distance indication claim 1 , and the method further comprises: 'processing the at ...

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.

INTERNET AUDIO SOFTWARE METHOD

An audio enhancement technique comprises an audio source. Presets for the input audio sound are provided. The presetted ausio sound is tone adjusted to create a tone adjusted presetted audio. The tone adjusted presetted audio is expanded to create an expanded audio and the expanded audio is outputted for use by user. The presetted audio is subjected to a treatment by one or more modules to improve sound quality. The treatment comprises processing the audio source by the following modules: a low pass filter with dynamic offset; an envelope controlled bandpass filter; a high pass filter; and adding an amount of dynamic synthesized sub bass to the audio. The processed audio signals are combined in a summing mixer with the audio source to create an audio out signal. In one embodiment, the expanding comprises sending audio out to a block that splits, processes and combines the stereo stream into several different versions that are fed into the Stereo Bus A; feeding Stereo Bus A output to a Compare Block that adjusts the amplitude of the original and processed audio by averaging; and feeding the output from the compare block as an stereo output. In one embodiment, the tone adjustment comprises: a first section for adjusting a low frequency tone; a second section for adjusting a mid frequency tone; a third section for adjusting a high frequency tone and mixing the audio outputs processed by the first, second and third sections to produce an output audio sound. 1. An audio enhancement technique comprising:an audio source;Setting presets for the input audio sound to create a presetted audio sound;Tone adjusting the presetted audio sound to create a tone adjusted presetted audio;Expanding the tone adjusted presetted audio to create an expanded audio;Outputting the expanded audio.2. The process of claim 1 , wherein the presetted audio is subjected to a treatment by one or more modules to improve sound quality.3. The process of claim 1 , wherein the treatment comprises ...

Method and apparatus for conversion of a multi-channel audio signal into a two-channel audio signal

A method for conversion of a n-channel audio signal (L, R, Ls, Rs) into a two-channel audio signal (Ro, Lo), where n≧4 and integer, includes the step of generating either one of the two-channel audio signals, right (Ro) or left (Lo), by a combination of: a front (R, L) and rear (Rs, Ls) signal components of the n-channel audio signal of the same side (right or left), and a front (L, R) signal component of the n-channel audio signal of the other side (left or right), and a term dependent of n.

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. 1. A method , comprising:receiving audio reproduction data comprising one or more audio objects and metadata associated with each of the one or more audio objects;receiving reproduction environment data comprising an indication of a number of reproduction speakers in the reproduction environment and an indication of the location of each reproduction speaker within the reproduction environment; andrendering the audio objects into one or more speaker feed signals by applying an amplitude panning process to each audio object, wherein the amplitude panning process is based, at least in part, on the metadata associated with each audio object and the location of each reproduction speaker within the reproduction environment, and wherein each speaker feed signal corresponds to at least one of the reproduction speakers within the reproduction environment;wherein the metadata associated with each audio object includes audio object coordinates indicating the intended reproduction position of the audio object within the reproduction environment and a snap flag indicating whether the amplitude panning process should render the audio object into a single speaker feed signal or apply panning rules to render the audio object into a plurality of speaker feed signals.2. The method of claim 1 , wherein:the snap flag indicates the amplitude panning process should render the audio object into a single speaker feed signal; andthe amplitude panning process renders the audio object into a ...

MULTICHANNEL SIGNAL PROCESSING METHOD, AND MULTICHANNEL SIGNAL PROCESSING APPARATUS FOR PERFORMING SAME

Provided are an encoding method of a multichannel signal, an encoding apparatus to perform the encoding method, a multichannel signal processing method, and a decoding apparatus to perform the decoding method. The decoding method may include identifying an N/2-channel downmix signal derived from an N-channel input signal; and generating an N-channel output signal from the identified N/2-channel downmix signal using a plurality of one-to-two (OTT) boxes. If a low frequency effect (LFE) channel is absent in the output signal, the number of OTT boxes may be equal to N/2 where N/2 denotes the number of channels of the downmix signal. 1. A multichannel signal processing method comprising:identifying an N/2-channel downmix signal derived from an N-channel input signal; andgenerating an N-channel output signal from the identified N/2-channel downmix signal using a plurality of one-to-two (OTT) boxes,wherein, if a low frequency effect (LFE) channel is absent in the output signal, the number of OTT boxes is equal to N/2 where N/2 denotes the number of channels of the downmix signal.2. The multichannel signal processing method of claim 1 , wherein each of the plurality of OTT boxes generates a 2-channel output signal using a 1-channel downmix signal and a decorrelated signal generated from a corresponding decorrelator.3. The multichannel signal processing method of claim 2 , wherein claim 2 , if N exceeds M where N denotes the number of channels of the output signal and M denotes the preset number of channels claim 2 , the decorrelator includes a first decorrelator corresponding to a channel of M or less and a second decorrelator corresponding to a channel greater than M claim 2 , andthe second decorrelator reuses a filter set of the first decorrelator.4. The multichannel signal processing method of claim 2 , wherein an OTT box from which an LFE channel is output claim 2 , among the plurality of OTT boxes claim 2 , generates a 2-channel downmix signal without using the ...

Reverberation Generation for Headphone Virtualization

The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well. 1. A method for generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization , comprising:generating directionally-controlled reflections that impart a desired perceptual cue to an audio input signal corresponding to a sound source location; andcombining at least the generated reflections to obtain the one or more components of the BRIR.2. The method of claim 1 , wherein the desired perceptual cues lead to a natural sense of space with minimal audible impairments.3. The method of claim 1 , wherein the directionally-controlled reflections have a predetermined direction of arrival in which an illusion of a virtual sound source at a given location in space is enhanced.4. The method of claim 3 , wherein the predetermined direction of arrival is of a wobble shape in which reflection directions change away from a virtual sound source and oscillate back and forth therearound.5. The method of claim 3 , wherein the predetermined direction of arrival further comprises a stochastic diffuse component within a predetermined azimuths range claim 3 , and wherein at least one of the wobble shapes or the stochastic diffuse components is selected based on a direction of the virtual sound source.6. The method of claim 1 , wherein generating directionally-controlled reflections comprises:determining respective occurrence time points of the ...

SIMULATING ACOUSTIC OUTPUT AT A LOCATION CORRESPONDING TO SOURCE POSITION DATA

Systems and methods of simulating acoustic output at a location corresponding to source position data are disclosed. A particular method includes receiving an audio signal and source position data associated with the audio signal. A set of speaker signals are applied to a plurality of speakers, where the set of speaker driver signals causes the plurality of speakers to generate acoustic output that simulates output of the audio signal by an audio source at a location corresponding to the source position data. 1. A method comprising:receiving, by an audio system in a vehicle, an audio signal and source position data associated with the audio signal;up-mixing the audio signal to generate a plurality of intermediate signal components;down-mixing the plurality of intermediate signal components to generate a plurality of speaker signal components; andprocessing a plurality of speaker signals components to generate a set of speaker driver signals that cause a plurality of speakers distributed within the vehicle to simulate output of the audio signal at a location corresponding to the source position data,wherein the plurality of speakers comprise a plurality of near-field speakers, and a plurality of fixed speakers located forward of the near-field speakers;wherein the set of speaker driver signals comprises a first plurality of speaker driver signals for delivery to the plurality of near-field speakers, and a second plurality of speaker driver signals for delivery to the plurality of fixed speakers located forward of the near-field speakers; and binaural filtering a subset of the plurality of speaker signal components to generate a plurality of binaural image signals;', 'filtering the plurality of binaural image signals by applying frequency response equalization of magnitude and phase to the plurality of binaural image signals;', 'combining the plurality of filtered binaural image signals to generate the first plurality of speaker driver signals; and', 'combining the ...

SPATIAL AUDIO SYSTEM FOR PLAYING LOCATION-AWARE DYNAMIC CONTENT

A method of presenting a soundscape in an environment space using a plurality of controllable speakers arranged the space is provided. A spatial audio engine may access a spatial audio file defining a digital representation of the soundscape, including (a) a plurality of sound elements to be output in the space over time, (b) sound playback location information defining location(s) for each sound element to be output, and (c) timing information defining time(s) for each sound element to be output. The spatial audio engine may also access speaker location information for each speaker in the space. For each sound element to be played in the space over time, the spatial audio engine may correlate the sound playback location information with the speaker location information to identify one or more of the speakers to be controlled to output audio signals associated with the respective sound element. 1. A method of presenting a scene using a plurality of controllable media output devices arranged in an environment space , the method comprising: a plurality of scene elements to be presented in the environment space over time;', 'scene element location information defining a location or range of locations for each scene element to be presented; and', 'timing information defining a time or range of times for each scene element to be presented;, 'accessing a digital representation of the scene to be presented in the environment space, the digital representation of the scene definingaccessing spatial information for each of the plurality of controllable media output devices; correlating the scene element location information for the respective scene element with the spatial information for the plurality of controllable media output devices to identify one or more of the controllable media output devices affected by the respective scene element; and', 'controlling the one or more controllable media output devices affected by the respective scene element to generate a human- ...

AV Timing Measurement and Correction for Digital Television

An invention for measuring, maintaining and correcting synchronization between signals which suffer varying relative delays during transmission and/or storage is shown. The present invention teaches measuring the relative delay between a plurality of signals which have suffered differing delays due to transmission, storage or other processing. The preferred embodiment of the invention includes the use of a marker which is generated in response to a second signal and combined with a first signal in a manner which ensures that the marker will not be lost in the expected processing of the first signal. Subsequently a first delayed marker is generated in response to the marker associated with or recovered from the first signal, and a second delayed marker is generated from the second signal. The first delayed marker and second delayed marker are compared to determine a measure of the relative timing or delay between said first signal and said second signal at said subsequent time. 1. An apparatus for generating a first set of markers in response to a relatively undelayed or delayed plurality of channels audio signal which is part of a high definition television program which first set of markers are intended to be utilized with a second set of markers generated in response to a delayed or undelayed version respectively of the audio signal , said apparatus including:a) an input circuit responsive to a digital audio portion of a high definition television program, said input circuit including combining a plurality of channels of said digital audio portion to provide at least a first digital output signal capable of carrying energy in a range of audio frequencies;b) an input filter circuit comprised of at least two digital filters, each responsive to said first digital output signals, each having a different amplitude response characteristic and each operative to provide a digital filter output signal having an amplitude which is responsive to one or more characteristic of ...

Audio apparatus and audio providing method thereof

Disclosed are an audio apparatus and an audio providing method thereof. The audio providing method includes receiving an audio signal including a plurality of channels, applying an audio signal having a channel, from among the plurality of channels, giving a sense of elevation to a filter to generate a plurality of virtual audio signals to be respectively output to a plurality of speakers, applying a combination gain value and a delay value to the plurality of virtual audio signals so that the plurality of virtual audio signals respectively output through the plurality of speakers form a sound field having a plane wave, and respectively outputting the plurality of virtual audio signals, to which the combination gain value and the delay value are applied, through the plurality of speakers. The filter processes the audio signal to have a sense of elevation.

METHOD FOR TRANSMITTING AND RECEIVING AUDIO DATA AND APPARATUS THEREFOR

A method for transmitting audio data performed by an audio data transmission apparatus in accordance with the present invention comprises the steps of: generating playback environment information of three-dimensional audio content; encoding a three-dimensional audio signal of the three-dimensional audio content; and transmitting, to an audio data reception apparatus, the encoded three-dimensional audio signal of the three-dimensional audio content and the generated playback environment information, wherein the playback environment information includes environment information of a room in which the three-dimensional audio content is played. 1. A method for transmitting audio data by an audio data transmission apparatus , the method comprising:generating playback environment information for three-dimensional (3D) audio content;encoding a 3D audio signal of the 3D audio content; andtransmitting the encoded 3D audio signal of the 3D audio content and the generated playback environment information to an audio data reception apparatus,wherein the playback environment information comprises environment information about a room for playing the 3D audio content.2. The method of claim 1 , wherein the environment information about the room for playing the 3D audio content comprises at least one of size information about the room or property information about the room.3. The method of claim 2 , wherein the size information about the room comprises at least one of information about a width of the room claim 2 , information about a length of the room claim 2 , or information about a height of the room.4. The method of claim 2 , wherein the property information about the room comprises at least one of a material property of a floor constituting the room claim 2 , a material property of a ceiling constituting the room claim 2 , a material property of a left wall constituting the room claim 2 , and a material property of a right wall constituting the room claim 2 , a material ...

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. 1. A method , comprising:receiving audio reproduction data comprising one or more audio objects and metadata associated with each of the one or more audio objects;receiving reproduction environment data comprising an indication of a number of reproduction speakers in the reproduction environment and an indication of the location of each reproduction speaker within the reproduction environment; andrendering the audio objects into one or more speaker feed signals by applying an amplitude panning process to each audio object, wherein the amplitude panning process is based, at least in part, on the metadata associated with each audio object and the location of each reproduction speaker within the reproduction environment, and wherein each speaker feed signal corresponds to at least one of the reproduction speakers within the reproduction environment;wherein the metadata associated with each audio object includes audio object coordinates indicating the intended reproduction position of the audio object within the reproduction environment and metadata indicating audio object spreads in two or more of three dimensions, wherein the audio object spreads are the same in the two or more dimensions, and wherein the rendering involves controlling the audio object spreads in the two or more dimensions in response to the metadata.2. An apparatus , comprising:an interface system; and receiving, via the interface system, audio reproduction data comprising one or more audio objects and ...

AUDIO SIGNAL PROCESSING METHOD

Disclosed is an audio signal processing method. The audio signal processing method according to the present invention comprises the steps of: receiving a bit-stream including at least one of a channel signal and an object signal; receiving a user's environment information; decoding at least one of the channel signal and the object signal on the basis of the received bit-stream; generating the user's reproducing channel information on the basis of the user's received environment information; and generating a reproducing signal through a flexible renderer on the basis of at least one of the channel signal and the object signal and the user's reproducing channel information. 1. An audio signal processing method performed by an audio signal processing device , comprising:receiving a bit-stream including at least one of a channel signal and an object signal;receiving user environment information;decoding at least one of the channel signal and the object signal based on the received bit-stream;generating a reproduction signal through a flexible renderer based on the user environment information and at least one of the channel signal and the object signal;determining gain and delay in consideration of information on at least one of speaker's position and user's position; andapplying the gain and delay to the reproduction signal,wherein the generating the reproduction signal generates a first reproduction signal in which the decoded channel signal and the decoded object signal are combined, using information about a user reproduction channel derived based on the user environment information,wherein generating the reproduction signal comprises:selecting three (3) channel signals that are adjacent to the object signal using position information of the object signal;multiplying the object signal by a gain value; andcombining the multiplied result with at least one of the selected channel signals, andwherein the gain value multiplied to the object signal is a VBAP gain value.2. ...

Statistical modelling, interpolation, measurement and anthropometry based prediction of head-related transfer functions

A system is disclosed for statistical modelling, interpolation, and user-feedback based inference of head-related transfer functions (HRTF) includes a processor performing operations that include using a collection of previously measured head related transfer functions for audio signals corresponding to multiple directions for at least one subject; and performing Gaussian process hyper-parameter training on the collection of audio signals. A method is disclosed for statistical modelling, interpolation, measurement and anthropometry based prediction of head-related transfer functions (HRTF) for a virtual audio system that includes collecting audio signals in transform domain for at least one subject; applying head related transfer functions (HRTF) measurement directions in multiple directions to the collected audio signals; and performing Gaussian hyper-parameter training on the collection of audio signals to generate at least one predicted HRTF. 1. A system for statistical modelling , interpolation , and user-feedback based inference of head-related transfer functions (HRTF) comprising: using a collection of previously measured head related transfer functions for audio signals corresponding to multiple directions for at least one subject; and', 'performing Gaussian process hyper-parameter training on the collection of audio signals., 'a tangible, non-transitory memory communicating with a processor, the tangible, non-transitory memory having instructions stored thereon that, in response to execution by the processor, cause the processor to perform operations comprising2. The system according to claim 1 , wherein the operation of performing Gaussian process hyper-parameter training on the collection of audio signals further comprises causing the processor to perform operations that include:applying sparse Gaussian process regression to perform the Gaussian process hyper-parameter training on the collection of audio signals.3. The system of claim 2 , further ...

Method for audio reproduction in a multi-channel sound system

The invention relates to a method for reproducing audio in a multi-channel sound system including two input signals (L and R), wherein output signals are generated for different sound perception levels. In order to develop said method in such a way that audio can be reproduced within a larger range of applications in a multi-channel sound system, according to the invention, only a lower sound perception level () and a higher sound perception level () are generated, and a maximum of six output signals are generated, a maximum of two output signals being allocated to the lower sound perception level () and a maximum of four output signals being allocated to the higher sound perception level (). 1. A device with sound input and sound output channels , as well as a processor , wherein loudspeakers are assigned to the device , wherein a software is imported onto the processor , which contains an algorithm , which is processed by the processor , wherein the algorithm covering a method for audio reproduction in a multi-channel sound system comprising two input signals L and R , wherein output signals are generated for different listening levels , wherein the device is modular with a first modular unit and one or more additional modular units , wherein in the first modular unit of the device only one lower listening level and only one upper listening level are generated , wherein a maximum of six output signals , with a maximum of two output signals for the lower listening level and a maximum of four output signals for the upper listening level , are generated , and wherein the first modular unit of the device is expanded in modular fashion in that the output signals of the first unit serve as input signals one of the one or more additional modular units , in order to generate further lower and upper listening levels by the one or more additional modular units of the device.2. The device according to claim 1 , wherein the first modular unit and the additional modular units ...

RENDERER CONTROLLED SPATIAL UPMIX

An audio decoder device for decoding a compressed input audio signal having at least one core decoder having one or more processors for generating a processor output signal based on a processor input signal, wherein a number of output channels of the processor output signal is higher than a number of input channels of the processor input signal, wherein each of the one or more processors has a decorrelator and a mixer, wherein a core decoder output signal having a plurality of channels has the processor output signal, and wherein the core decoder output signal is suitable for a reference loudspeaker setup; at least one format converter device configured to convert the core decoder output signal into an output audio signal, which is suitable for a target loudspeaker setup; and a control device configured to control at least one or more processors in such way that the decorrelator of the processor may be controlled independently from the mixer of the processor, wherein the control device is configured to control at least one of the decorrelators of the one or more processors depending on the target loudspeaker setup.

MULTIDIMENSIONAL VIRTUAL LEARNING SYSTEM AND METHOD

A process and system for generating three dimensional sound conferencing includes generating a virtual map with a plurality of positions, each participant selecting one of the positions, determining a direction from each position to each other position on the map, determining a distance from each position to each other position on the map, receiving sound from each participant, mixing the received sound, transforming the mixed sound into binaural audio, and directing the binaural audio sound to each participant via a speaker associated with the virtual position of the speaking participant. The result is a clarified sound that gives to the listening participant a sense of where the speaking participant is positioned relative to the listening participant. 1. A computer implemented multi-dimensional sound conferencing method for a plurality of conference participants comprising:assigning, via a processor, each conference participant to a unique position on a computer generated map of one of a real or virtual venue; wherein the plurality of conference participants includes speaking participants and listening participants;determining a direction on the map of each participant from each other participant;receiving a voice sound from one of the speaking participants;leveling the received voice sound from the one speaking participant to a common leveled voice sound for the speaking participant;converting the leveled voice sound to a converted sound corresponding to the direction to each of the listening participants from the one speaking participant;transforming the converted sound to binaural sound for transmission to each listening participant; andafter the transforming step, canceling a crosstalk effect in the binaural sound such that the binaural sound is configured to maintain accurate directional information when a listening participant is not using headphones.2. A method according to wherein the map is a virtual room.3. A method according to wherein the virtual room ...

Reverberation Generation for Headphone Virtualization

The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well.

UPMIXING OF AUDIO SIGNALS

Example embodiments disclosed herein relates to upmixing of audio signals. A method of upmixing an audio signal is described. The method includes decomposing the audio signal into a diffuse signal and a direct signal, generating an audio bed at least in part based on the diffuse signal, the audio bed including a height channel, extracting an audio object from the direct signal, estimating metadata of the audio object, the metadata including height information of the audio object; and rendering the audio bed and the audio object as an upmixed audio signal, wherein the audio bed is rendered to a predefined position and the audio object is rendered according to the metadata. Corresponding system and computer program product are described as well. 1. A method of upmixing an audio signal , comprising:decomposing the audio signal into a diffuse signal and a direct signal;generating an audio bed at least in part based on the diffuse signal, the audio bed including a height channel;extracting an audio object from the direct signal;estimating metadata of the audio object, the metadata including height information of the audio object; andrendering the audio bed and the audio object as an upmixed audio signal, wherein the audio bed is rendered to a predefined position and the audio object is rendered according to the metadata.2. The method of claim 1 , wherein the generating the audio bed comprises:upmixing the diffuse signal to create the height channel; andincluding a residual signal into the audio bed, the residual signal obtained from the extracting of the audio object.3. The method of claim 1 , wherein the decomposing the audio signal comprises:applying a first decomposition process to obtain the diffuse signal; andapplying a second decomposition process to obtain the direct signal, the first decomposition process having less diffuse-to-direct leakage than the second decomposition process.4. The method of claim 3 , further comprising:pre-upmixing the audio signal,wherein ...

METHOD AND APPARATUS FOR CHANGING THE RELATIVE POSITIONS OF SOUND OBJECTS CONTAINED WITHIN A HIGHER-ORDER AMBISONICS REPRESENTATION

Higher-order Ambisonics HOA is a representation of spatial sound fields that facilitates capturing, manipulating, recording, transmission and playback of complex audio scenes with superior spatial resolution, both in 2D and 3D. The sound field is approximated at and around a reference point in space by a Fourier-Bessel series. The invention uses space warping for modifying the spatial content and/or the reproduction of sound-field information that has been captured or produced as a higher-order Ambisonics representation. Different warping characteristics are feasible for 2D and 3D sound fields. The warping is performed in space domain without performing scene analysis or decomposition. Input HOA coefficients with a given order are decoded to the weights or input signals of regularly positioned (virtual) loudspeakers. 110-. (canceled)11. Method for changing the relative positions of sound objects contained within a two-dimensional or a three-dimensional Higher-Order Ambisonics HOA representation of an audio scene , wherein an input vector Awith dimension Odetermines the coefficients of a Fourier series of the input signal and an output vector Awith dimension Odetermines the coefficients of a Fourier series of the correspondingly changed output signal , said method comprising the steps:{'sub': in', 'in', '1', '1', 'in', '1', 'in, 'sup': −1', '−1, 'decoding said input vector Aof input HOA coefficients into input signals sin space domain for regularly positioned loudspeaker positions using the inverse Ψof a mode matrix Ψby calculating s=ΨA;'}{'sub': in', 'out', 'out', '2', 'in', '2', 'out, 'warping and encoding in space domain said input signals sinto said output vector Aof adapted output HOA coefficients by calculating A=Ψs, wherein the mode vectors of the mode matrix Ψare modified according to a warping function ƒ(φ) by which the angles of the original loudspeaker positions are one-to-one mapped into the target angles of the target loudspeaker positions in said output ...

Manipulation of Playback Device Response Using Signal Processing

An example playback device (i) provides a center channel of audio content to one or more first audio drivers and one or more second audio drivers so that the center channel is reproduced according to a first radiation pattern and (ii) provides a side channel of audio content to the one or more first audio drivers so that the side channel is reproduced according to a second radiation pattern. The first and second radiation patterns may combine to form a response lobe that has a maximum between the respective maxima of the first and second radiation patterns. An example non-transitory computer-readable medium and an example method, both related to the example playback device, are also disclosed herein. 1. A playback device comprising:one or more processors;one or more first audio drivers;one or more second audio drivers; and receiving a left channel of audio content and a right channel of the audio content;', 'generating a center channel of the audio content comprising a combination of the left and right channels;', 'providing the generated center channel to (i) the one or more first audio drivers and (ii) the one or more second audio drivers for playback of the center channel according to a first radiation pattern that has a maximum along a first direction;', 'generating a side channel comprising a combination of (i) the center channel and (ii) a difference between the left channel and the right channel; and', 'providing the generated side channel to the one or more first audio drivers for playback of the side channel according to a second radiation pattern that has a maximum along a second direction, wherein the first radiation pattern and the second radiation pattern combine to form a first response lobe that has a maximum along a third direction between the first and second directions., 'a non-transitory computer-readable medium storing instructions that, when executed by the one or more processors, cause the playback device to perform functions comprising2. The ...

SYSTEM, METHOD, AND APPARATUS FOR GENERATING AND DIGITALLY PROCESSING A HEAD RELATED AUDIO TRANSFER FUNCTION

The present invention provides for an apparatus, system, and method for generating a head related audio transfer function in real time. Specifically, the present invention utilizes unique structural components including a tragus structure and an antihelix structure in connection with a microphone in order to communicate the location of a sound in three dimensional space to a user. The invention also utilizes an audio processor to digitally process the head related audio transfer function. The system may also be utilized to pan the directionality of audio sources within a virtual environment at least partially in response to movement of a user. 1. A system for generating an adjusted head related audio transfer function (HRTF) for a user , said system comprising:at least one input signal containing information corresponding to an origin angle of said input signal,at least one audio processor configured to calculate an output based on said origin angle,a motion translator configured to receive motion input corresponding to the motion of a user's head and modify said at least one origin angle according to said motion input,an array providing predetermined decibel levels for an output signal based in part on said origin angle and said motion input,a plurality of signal filters configured to receive and filter said output signal based at least in part on said array,a plurality of speakers in communication with the signal filters structured and disposed to relay said output signal.2. The system of wherein the plurality of signal filters comprises a plurality of FIR filter pairs.3. The system of wherein said plurality of FIR filter pairs are arranged in a Mid-Side configuration.4. The system of wherein said plurality of FIR filter pairs are configured with a left input channel and a right input channel.5. The system of wherein said at least one audio processor comprises a one-to-many upmixer configured to process a number of audio channels into a greater number of audio ...

SYSTEMS AND METHODS FOR SIMULATION OF MIXING IN AIR OF RECORDED SOUNDS

A mixing signal processing technique modifies digital audio recordings to simulate the linear and nonlinear effects of propagation and mixing of sounds in air. When multiple sounds or complex sounds comprised of multiple frequencies in the audible spectrum propagate in such a nonlinear medium, they transfer energy into sound at new frequencies given by the sums and differences of the original signal frequencies. The mixing signal processing technique may improve the ability of a system to reproduce the effects of a live performance using a digital audio recording. 1. A method for simulating propagation and mixing of sounds in air , the method comprising:selecting a relationship between a virtual audio listener location and a virtual source location within an audio interaction volume, the audio interaction volume associated with a first audio waveform, the first audio waveform including audio generated using a first audio isotropic source; andgenerating a second audio waveform using the first audio waveform, the second audio waveform including a plurality of simulated intermodulation products corresponding to the relationship between the virtual audio listener location and the virtual source location.2. The method of claim 1 , the first audio waveform further including audio generated using a second isotropic audio source.3. The method of claim 1 , further including generating a simulated mixed output waveform claim 1 , the generating including:applying a first gain to the first audio waveform to generate a first amplified waveform;applying a second gain to the second audio waveform to generate a second amplified waveform; andsumming the first amplified waveform and the second amplified waveform to generate the simulated mixed output waveform.4. The method of claim 3 , further including transducing the simulated mixed output waveform into audible sounds.5. The method of claim 1 , further including:identifying an audio sample within the second audio waveform; ...

APPARATUS AND METHOD FOR SEPARATING MULTI-CHANNEL AUDIO SIGNAL

An apparatus and method for separating a multi-channel audio signal that separates a multi-channel audio signal into a plurality of sound source objects is disclosed, the apparatus including a multi-channel stereo transformer to transform a multi-channel audio signal into a plurality of stereo signals, and a stereo sound source separator to separate the plurality of stereo signals into a plurality of sound source objects. 1. An apparatus for separating a multi-channel audio signal , the apparatus comprising:a multi-channel-stereo transformer to transform a multi-channel audio signal into a plurality of stereo signals; anda stereo sound source separator to separate the plurality of stereo signals into a plurality of sound source objects.2. The apparatus of claim 1 , wherein the multi-channel-stereo transformer transforms the multi-channel audio signal into a signal of a time-frequency region claim 1 , and transforms the multi-channel audio signal into the plurality of stereo signals through use of a cross correlation coefficient of a time-frequency (TF) bin.3. The apparatus of claim 2 , wherein the multi-channel-stereo transformer determines a mask to be applied to the multi-channel audio signal transformed into the time-frequency region based on the cross correlation coefficient claim 2 , and generates a stereo signal through use of the determined mask.5. The apparatus of claim 1 , wherein the multi-channel-stereo transformer comprises:a time-frequency transformer to transform the multi-channel audio signal into a time-frequency region;a cross correlation coefficient calculator to calculate a cross correlation coefficient of a TF bin in the multi-channel audio signal transformed into the time-frequency region;a mask determiner to determine a mask to be applied to the multi-channel audio signal transformed into the time-frequency region based on the cross correlation coefficient; anda stereo signal generator to generate a stereo signal through use of the mask.6. The ...

AUDIO RENDERING SYSTEM CATEGORISING GEOSPATIAL OBJECTS

The present disclosure relates to a method and an audio rendering system comprising at least one portable terminal configured to receive geospatial object data from at least one geospatial object data server. The geospatial object data is interrelated to a geographical position. The at least one portable terminal is further configured to render retrieved geospatial object data into an acoustic scene by a rendering algorithm. The acoustic scene is spatially interrelated to the geographical position. The at least one audio unit is configured to sound a rendered acoustic scene information into at least one ear of a user. The audio rendering system is further configured for rendering retrieved geospatial object data into the acoustic scene based on categorised acoustic scene information representing a corresponding categorised geospatial object data. 1. An audio rendering system comprising:at least one portable terminal configured to receive geospatial object data from at least one geospatial object data server, the geospatial object data being interrelated to a geographical position, the at least one portable terminal being configured to render the retrieved geospatial object data into an acoustic scene using a rendering algorithm, the acoustic scene being spatially interrelated to the geographical position in such a way that the acoustic scene is perceived observed from the geographical position; andat least one audio unit configured to sound a rendered acoustic scene information into at least one ear of a user;wherein the at least one portable terminal is configured to render the retrieved geospatial object data into the acoustic scene based on categorized acoustic scene information representing corresponding categorized geospatial object data.2. The audio rendering system according to claim 1 , wherein the categorized acoustic scene information comprises a distinguishing sound representing the corresponding categorized geospatial object data.3. The audio rendering ...

METHOD AND APPARATUS FOR REPRODUCING THREE-DIMENSIONAL AUDIO

A three-dimensional (3D) audio reproducing method and apparatus is provided. The 3D audio reproducing method may include receiving a multichannel signal comprising a plurality of input channels; and performing downmixing according to a frequency range of the multichannel signal in order to format-convert the plurality of input channels into a plurality of output channels having elevation. 1. A method of rendering an audio signal , the method comprising:receiving a plurality of input channel signals including a height input channel signal;generating a parameter for phase-aligning based on the plurality of input channel signals;modifying a downmix matrix, based on the parameter for phase-aligning, to phase-align a first frequency range of the plurality of input channel signals; anddownmixing the plurality of input channel signals to a plurality of output channel signals based on the modified downmix matrix,wherein the first frequency range includes below 2.8 kHz and above 10 kHz,wherein the height input channel signal is identified based on elevation information, andwherein the modified downmix matrix includes two types comprising a first downmix matrix for a general scene and a second downmix matrix for a highly decorrelated wideband scene, and the downmixing is performed by one of the first downmix matrix or the second downmix matrix selected according to a received flag.2. An apparatus for rendering an audio signal , the apparatus comprising:a processor; anda memory storing instructions executable by the processor,wherein the processor is configured to:receive a plurality of input channel signals including a height input channel signal;generate a parameter for phase-aligning based on the plurality of input channel signals;modify a downmix matrix, based on the parameter for phase-aligning, to phase-align a first frequency range of the plurality of input channel signals; anddownmix the plurality of input channel signals to a plurality of output channel signals based ...

Parametric stereo upmix apparatus, a parametric stereo decoder, a parametric stereo downmix apparatus, a parametric stereo encoder

A parametric stereo upmix method for generating a left signal and a right signal from a mono downmix signal based on spatial parameters includes predicting a difference signal comprising a difference between the left signal and the right signal based on the mono downmix signal scaled with a prediction coefficient. The prediction coefficient is derived from the spatial parameters. The method further includes deriving the left signal and the right signal based on a sum and a difference of the mono downmix signal and said difference signal.

Apparatus and method for generating multimedia data, and apparatus and method for playing multimedia data

An apparatus and method for generating multimedia data and an apparatus and method for playing multimedia data are disclosed. The apparatus for generating the multimedia data may include a spatial information identification unit to identify spatial information for a plurality of channels of a multi-channel audio signal, and a multimedia data generation unit to generate multimedia data including the spatial information, and the apparatus for playing the multimedia data may include a spatial information analysis unit to analyze spatial information for a plurality of channels of a multi-channel audio signal included in multimedia data, and a multimedia data playback unit to play multimedia data, based on the spatial information.

Location information through directional sound provided by mobile computing device

A mobile computing device that provides location information through directional sound is described herein. The mobile computing device includes a location detection system that provides location signals corresponding to a user location and a destination location, such as a vehicle location or a vertex of a predefined travel route, to a spatial audio generation system to define a spatial audio signal based on a direction from the user location to the destination location. The spatial audio signal is provided to an audio device of the mobile computing device that outputs the spatial audio signal as directional sound having a locus at the destination location.

Headphone Device for Reproducing Three-Dimensional Sound Therein, and Associated Method

3D audio virtualization within headphone-type sound reproduction devices, comprises: deriving an HRTF, comprising a PRTF, that includes acoustical effects due to pinnae and ear canals, and a remainder HRTF, that includes acoustical effects due to head, shoulders, torso and other body parts while excluding acoustical effects from pinnae and ear canals; wherein the remainder HRTF is electronically implemented and omits acoustical effects due to pinnae and ear canal effects; and wherein the PRTF is acoustically implemented and personalized to the user through use of two or more transducers positioned such that a front plane of the transducer, the front plane of the transducer's diaphragm, the transducer's mechanical center or the transducer's acoustical center point are 25 mm or more from a user's ear canal entrance, and/or oriented so the 0° axis of acoustical output is aligned with the acoustical output axes of typical external loudspeakers positioned in the acoustical far-field. 1. A method for providing 3D audio virtualization within headphone-type sound reproduction devices , comprising the steps of:deriving a composite Head-Related Transfer Function (HRTF) consisting of a cascade, or series combination, comprising a Pinna-Related Transfer Function (PRTF), that includes the acoustical effects due to pinnae and ear canals, and a remainder HRTF, that includes acoustical effects due to head, shoulders, torso and other body parts while excluding acoustical effects due to pinnae and ear canals;wherein the remainder HRTF is electronically implemented using either digital processing or analog processing, and omits the acoustical effects due to pinnae and ear canal effects; andwherein the PRTF component is acoustically implemented and personalized to the user through the use of two or more transducers that are positioned such that a front plane of the transducer, the front plane of the transducer's diaphragm, the transducer's mechanical center or the transducer's ...

APPARATUS AND METHOD FOR EDGE FADING AMPLITUDE PANNING

An apparatus for generating four or more audio output signals has a panning gain determiner and a signal processor. The panning gain determiner is configured to determine a proper subset from a set of five or more loudspeaker positions, so that the proper subset has four or more of the five or more loudspeaker positions. Moreover, the panning gain determiner is configured to determine the proper subset depending on a panning position and on the five or more loudspeaker positions, and to determine a panning gain for each of the four or more audio output signals by determining the panning gain depending on the panning position and on the four or more loudspeaker positions of the proper subset. The signal processor is configured to generate each of the four or more audio output signals depending on the panning gain for the audio output signal and on an audio input signal. 1. An apparatus for generating four or more audio output signals , comprising:a panning gain determiner, anda signal processor,wherein the panning gain determiner is configured to determine a proper subset from a set of five or more loudspeaker positions, so that the proper subset comprises four or more of the five or more loudspeaker positions,wherein the panning gain determiner is configured to determine the proper subset depending on a panning position and depending on the five or more loudspeaker positions,wherein the panning gain determiner is configured to determine a panning gain for each of the four or more audio output signals by determining said panning gain depending on the panning position and depending on the four or more loudspeaker positions of the proper subset, andwherein the signal processor is configured to generate each audio output signal of the four or more audio output signals depending on the panning gain for said audio output signal and depending on an audio input signal.2. The apparatus according to claim 1 ,wherein each loudspeaker position of the four or more loudspeaker ...

STEREO EXTENSION APPARATUS AND METHOD

Disclosed herein are a stereo extension apparatus and method. The apparatus includes a database that stores predetermined information as a result of Gaussian mixture model (GMM) training or hidden Markov model (HMM) training; a modified discrete cosine transform (MDCT) transformer that transforms a mono signal through MDCT, a feature parameter extractor that extracts a feature parameter of the mono signal from an MDCT coefficient output from the MDCT transformer, a side signal energy estimator that estimates subband energy of a side signal with reference to information stored in the database based on the feature parameter; an energy controller that obtains the MDCT coefficient of a side signal estimated from the subband energy of the estimated side signal, an inverse MDCT transformer that obtains an estimated side signal by transforming the MDCT coefficient of the estimated side signal through inverse MDCT. 1. A stereo extension apparatus comprising:a database that stores predetermined information as a result of Gaussian mixture model (GMM) training or hidden Markov model (HMM) training;a modified discrete cosine transform (MDCT) transformer that transforms a mono signal through MDCT;a feature parameter extractor that extracts a feature parameter of the mono signal from an MDCT coefficient output from the MDCT transformer;a side signal energy estimator that estimates subband energy of a side signal with reference to information stored in the database based on the feature parameter;an energy controller that obtains the MDCT coefficient of a side signal estimated from the subband energy of the estimated side signal;an inverse MDCT transformer that obtains an estimated side signal by transforming the MDCT coefficient of the estimated side signal through inverse MDCT; anda stereo signal generator that obtains a stereo signal based on sum and difference between the mono signal and the estimated side signal.2. The stereo extension apparatus according to claim 1 , further ...

Audio Processing Method and Audio Processing Apparatus

An audio processing method and an audio processing apparatus are described. A mono-channel audio signal is transformed into a plurality of first subband signals. Proportions of a desired component and a noise component are estimated in each of the subband signals. Second subband signals corresponding respectively to a plurality of channels are generated from each of the first subband signals. Each of the second subband signals comprises a first component and a second component obtained by assigning a spatial hearing property and a perceptual hearing property different from the spatial hearing property to the desired component and the noise component in the corresponding first subband signal respectively, based on a multi-dimensional auditory presentation method. The second subband signals are transformed into signals for rendering with the multi-dimensional auditory presentation method. By assigning different hearing properties to desired sound and noise, the intelligibility of the audio signal can be improved. 130-. (canceled)31. An audio processing method comprising:transforming a mono-channel audio signal into a plurality of first subband signals;estimating proportions of a desired component and a noise component in each of the subband signals;generating second subband signals corresponding respectively to a plurality of channels from each of the first subband signals, wherein each of the second subband signals comprises a first component and a second component obtained by assigning a spatial hearing property and a perceptual hearing property different from the spatial hearing property to the desired component and the noise component in the corresponding first subband signal respectively, based on a multi-dimensional auditory presentation method; andtransforming the second subband signals into signals for rendering with the multi-dimensional auditory presentation method.32. The audio processing method according to claim 31 , wherein generating second subband ...

MULTIMEDIA AUDIO INSERTER

In one embodiment, a multimedia audio inserter removes the left and right audio channel data from a multimedia signal, for example an HDMI® signal and replaces it with an alternate audio signal, which may be the center channel audio signal provided by a digital surround sound receiver. In exemplary applications, the resulting, modified multimedia signal may be provided to a digital television that is part of a surround sound system, enabling the television to play the center channel signal through its speakers. Optionally, the multimedia audio inserter may provide a copy of the multimedia signal at a multimedia feedthrough output. In exemplary applications, the multimedia feedthrough output may be used to play left and right channel audio, present in the HDMI® signal, through dedicated left and right speakers. 1. A multimedia audio inserter comprising:a de-audio device configured to receive a first multimedia input signal, separate the first multimedia input signal into one or more audio signals and one or more video signals, and to output one or more de-audio signals, the de-audio signals comprising the video signals and fewer than all of the audio signals; andan audio injection device coupled to the output of the de-audio device and configured to receive alternate audio signals and combine them with the one or more de-audio signals to form a modified multimedia signal.2. The multimedia audio inserter of claim 1 , further comprising:a multimedia input device configured to receive an external multimedia signal and to output the first multimedia input signal and a second multimedia input signal, both the first and the second multimedia input signals being identical to the external multimedia signal;a multimedia output device configured to communicate the modified multimedia signal to a first audiovisual display device; anda multimedia feedthrough output, configured to communicate the second multimedia input signal to a second audiovisual display device.3. The ...

SOUND PROCESSING APPARATUS AND METHOD, AND PROGRAM

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

FAST AND MEMORY EFFICIENT ENCODING OF SOUND OBJECTS USING SPHERICAL HARMONIC SYMMETRIES

A method of encoding sound objects includes receiving a set of monophonic sound inputs. Each of the set of monophonic sound inputs includes position and orientation information of a sound object relative to a source position. The set of monophonic sound inputs are encoded into a higher order ambisonic (HOA) sound field in a spherical harmonics domain based on a spherical harmonics dataset including a subset of all spherical harmonic coefficients for a given subset of azimuth and elevation angles. Some embodiments include decoding the HOA sound field to generate a set of loudspeaker signals. 1. A method , comprising:receiving a set of monophonic sound inputs, wherein each of the set of monophonic sound inputs includes position and orientation information of a sound object relative to a source position; andencoding, based on a spherical harmonics dataset including a subset of all spherical harmonic coefficients for a given subset of azimuth and elevation angles, the set of monophonic sound inputs into a higher order ambisonic (HOA) sound field in a spherical harmonics domain; anddecoding the HOA sound field to generate a set of loudspeaker signals.2. The method of claim 1 , further comprising:convolving two or more of the generated set of loudspeaker signals and summing a left channel signal and a right channel signal to generate a binaural sound output.3. The method of claim 1 , further comprising:storing the spherical harmonics dataset as a spherical harmonics lookup table.4. The method of claim 3 , further comprising:computing the spherical harmonics lookup table prior to performing sound object encoding.5. The method of claim 4 , wherein computing the spherical harmonics lookup table comprises computing a first sphere quadrant of all spherical harmonic coefficients.6. The method of claim 5 , further comprising:retrieving, based on a symmetry look-up table, a spherical harmonic coefficient of a second sphere quadrant from the spherical harmonics lookup table.7. The ...

SYSTEM AND METHOD FOR ENHANCING VIRTUAL AUDIO HEIGHT PERCEPTION

An audio processing system for enhancing a user's virtual audio height perception, comprising: a rebalancing module for receiving audio signals from a source, the audio signals including a low layer signal representing sounds for transmission directly towards a user, and a height signal representing sounds for transmission towards the user by reflecting off a predetermined location above the user; the rebalancing module for comparing the height signal and the lower layer signal, and adjusting an amplitude of the height signal based on said comparison; the low layer signal being transmitted to one or more speakers of a first speaker arrangement for transmitting sound represented by the lower layer signal directly towards the user; and the adjusted height signal being transmitted to one or more speakers of a second speaker arrangement for transmitting sound represented by the height signal towards the user by reflecting off the predetermined location above the user. 1. An audio processing system for enhancing a user's virtual audio height perception , comprising:a rebalancing module for receiving audio signals from a source, the audio signals including a low layer signal representing sounds for transmission directly towards a user, and a height signal representing sounds for transmission towards the user by reflecting off a predetermined location above the user;the rebalancing module for comparing the height signal and the low layer signal, and adjusting an amplitude of the height signal based on said comparison;the low layer signal being transmitted to one or more speakers of a first speaker arrangement for transmitting sound represented by the lower layer signal directly towards the user; andthe adjusted height signal being transmitted to one or more speakers of a second speaker arrangement for transmitting sound represented by the height signal towards the user by reflecting off the predetermined location above the user.2. An audio processing system according to ...

METHOD AND APPARATUS FOR A BINAURAL HEARING ASSISTANCE SYSTEM USING MONAURAL AUDIO SIGNALS

The present application provides method and apparatus for a binaural hearing assistance system using a monaural audio signal input. The system, in various examples, provides adjustable delay/phase adjustment and sound level adjustment. Different embodiments are provided for receiving the monaural signal and distributing it to a plurality of hearing assistance devices. Different relaying modes are provided. Special functions are supported, such as telecoil functions. The system also has examples that account for a head-related transfer function in providing advanced sound processing for the wearer. Other examples are provided that are described in the detailed description. 1. (canceled)2. A method for providing a sound to a listener having first and second ears , the method comprising:receiving a monaural information signal;converting the monaural information signal into first and second output signals using a processor, including introducing a phase difference between the first and second output signals;receiving a control signal;adjusting the phase difference using the control signal; and providing a first sound based on the first output signal to the first ear using a first speaker; and', 'providing a second sound based on the second output signal to the second ear using a second speaker., 'providing the sound to the listener to be perceived by the listener as a binaural sound having an interaural difference, including3. The method of claim 2 , further comprising varying the phase difference as a function of frequency.4. The method of claim 2 , further comprising adjusting the phase difference using time-domain filters.5. The method of claim 2 , further comprising adjusting the phase difference based on a head-related transfer function.6. The method of claim 2 , wherein receiving the control signal comprises receiving the control signal from one or more switches operable by the listener.7. The method of claim 2 , wherein converting the monaural information signal ...

SPATIALLY REPRESENTING GRAPHICAL INTERFACE ELEMENTS AS BINAURAL AUDIO CONTENT

Certain embodiments involve spatially representing graphical interface elements as binaural audio content. For example, a system accesses electronic content having various content elements and identifies the various content elements. The system generates a three-dimensional audio representation for a content element by: accessing an audio representation of the content element; identifying a visual location of the content element; and generating a three-dimensional audio representation of the content element by applying, to the audio representation of the content element and using the visual location, a finite impulse response filter using a head-related transfer function, wherein the head-related transfer function comprises a set of filter parameters that model the visual location of the content element and wherein, when played, the three-dimensional audio representation appears to originate from the visual location. The system can then output the three-dimensional audio representation. 1. A method for spatially representing graphical interface elements as binaural audio content , the method comprising:accessing, by a processor, electronic content having a plurality of content elements;identifying, by the processor, the plurality of content elements of the electronic content; accessing an audio representation of the content element;', accessing, by the processor, a content object model specifying visual locations of the plurality of content elements when displayed in a graphical interface; and', 'identifying, by the processor and from the content object model, the visual location of the content element; and, 'identifying a visual location of the content element by;'}, 'generating a three-dimensional audio representation of the content element by applying, to the audio representation of the content element and using the visual location, a finite impulse response filter using a head-related transfer function, wherein the head-related transfer function comprises a set ...

Sound Stage Controller for a Near-Field Speaker-Based Audio System

Signals in an automobile audio system having at least two near-field speakers located close to an intended position of a listener's head are adjusted such that in a first mode, audio signals are distributed to the near-field speakers according to a first filter that causes the listener to perceive a wide soundstage, and in a second mode, the audio signals are distributed to the near-field speakers according to a second filter that causes the listener to perceive a narrow soundstage. A user input of a variable value is received and, in response, distribution of the audio signals is transitioned from the first mode to the second mode, the extent of the transition being variable based on the value of the user input. 1. A computer-implemented method of adjusting signals in an automobile audio system having at least two near-field speakers located close to an intended position of a listener's head in a vehicle cabin of an automobile and at least a first fixed speaker and a second fixed speaker positioned near a left corner and a right corner , respectively , of the vehicle cabin forward of the intended position of the listener's head , the computer-implemented method comprising: identifying content of the source audio signal as being of a first type or a second type; and', 'upmixing the source audio signal to generate at least three component channel signals;, 'processing a source audio signal having at least two channels, includingif the content is identified as being of the first type, applying a first set of weights to the component channel signals at each of a set of designated positions other than the actual locations of the speakers to cause the listener to perceive a first soundstage, otherwise, applying a second set of weights to the component channel signals at each of the set of designated positions to cause the listener to perceive a second sound stage;receiving user input to effect a transition between the first soundstage and the second soundstage to a user- ...

Method for Playing Back the Sound of a Digital Audio Signal

A method for playing back the sound of a digital audio signal comprising an oversampling step consisting of producing, from a signal sampled at a frequency F, a signal sampled at a frequency N×F, where N corresponds to an integer greater than 1, then of applying convolution processing to a first digital file sampled at a frequency N×F corresponding to the acquisition of the soundscape of a reference sound space, a second digital file sampled at a frequency N×F corresponding to the acquisition of the noise footprint of a piece of reference playback equipment, a third digital file sampled at a frequency N×F corresponding to the acquisition of the noise footprint of an equalizer and a fourth file corresponding to said oversampled audio file, the resulting digital packets then undergoing digital conversion processing at a sampling frequency F/M corresponding to the working frequency of the listening equipment. 12-. (canceled)3. A method for playing back the sound of a digital audio signal , comprising: a step of oversampling is executed , which consists in producing , from a signal sampled at a frequency F , a signal sampled at a frequency N×F , where N corresponds to an integer greater than 1 , then in applying convolution processing to a first digital file sampled at a frequency N×F corresponding to the acquisition of the soundscape of a reference sound space , a second digital file sampled at a frequency N×F corresponding to the acquisition of the noise footprint of a piece of reference playback equipment , a third digital file sampled at a frequency N×F corresponding to the acquisition of the noise footprint of an equalizer and a fourth file corresponding to said oversampled audio file , with the resulting digital packets then undergoing digital conversion processing at a sampling frequency F/M corresponding to the working frequency of the listening equipment.4. The method for playing back the sound of a digital audio signal according to claim 3 , further comprising ...

SOUND PROCESSING APPARATUS AND SOUND PROCESSING SYSTEM

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

AUDIO SIGNAL PROCESSING METHOD

Disclosed is an audio signal processing method. The audio signal processing method according to the present invention comprises the steps of: receiving a bit-stream including at least one of a channel signal and an object signal; receiving a user's environment information; decoding at least one of the channel signal and the object signal on the basis of the received bit-stream; generating the user's reproducing channel information on the basis of the user's received environment information; and generating a reproducing signal through a flexible renderer on the basis of at least one of the channel signal and the object signal and the user's reproducing channel information. 1. An audio signal processing method , comprising:receiving a bit-stream including at least one of a channel signal and an object signal;receiving user environment information;decoding at least one of the channel signal and the object signal based on the received bit-stream;generating user reproduction channel information using the received user environment information; andgenerating a reproduction signal through a flexible renderer based on the user reproduction channel information and at least one of the channel signal and the object signal.2. The audio signal processing method of claim 1 , wherein generating the user reproduction channel information determines whether a number of the user reproduction channels is identical to a number of channels of a standard specification claim 1 , based on the received user environment information.3. The audio signal processing method of claim 2 , wherein when the number of the user reproduction channels is identical to the number of channels of the standard specification claim 2 , the decoded object signal is rendered according to the number of the user reproduction channels claim 2 , andwhen the number of the user reproduction channels is not identical to the number of channels of the standard specification, the decoded object signal is rendered in response ...

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. 1. A method , comprising:receiving audio reproduction data comprising one or more audio objects and metadata associated with each of the one or more audio objects;receiving reproduction environment data comprising an indication of a number of reproduction speakers in the reproduction environment and an indication of the location of each reproduction speaker within the reproduction environment; andrendering the audio objects into one or more speaker feed signals by applying an amplitude panning process to each audio object, wherein the amplitude panning process is based, at least in part, on the metadata associated with each audio object and the location of each reproduction speaker within the reproduction environment, and wherein each speaker feed signal corresponds to at least one of the reproduction speakers within the reproduction environment;wherein the metadata associated with each audio object includes audio object coordinates indicating the intended reproduction position of the audio object within the reproduction environment and metadata indicating an audio object spread in one or more of three dimensions, and wherein the rendering involves controlling the audio object spread in the one or more dimensions in response to the metadata.2. The method of claim 1 , wherein the metadata indicates audio object spreads in two or more of the three dimensions claim 1 , and wherein the rendering involves controlling the audio object spread in the two or more dimensions in ...

SYSTEM, APPARATUS AND METHOD FOR CONSISTENT ACOUSTIC SCENE REPRODUCTION BASED ON INFORMED SPATIAL FILTERING

A system for generating and outputting one or more audio output signals has a decomposition module, a signal processor, and an output interface. The decomposition module can receive two or more audio input signals, to generate a direct component signal, having direct signal components of the audio input signals, and to generate a diffuse component signal, having diffuse signal components of the audio input signals. The signal processor can receive the direct component signal, the diffuse component signal and direction information, to generate one or more processed diffuse signals depending on the diffuse component signal, for each of the one or more audio output signals, to determine, depending on the direction of arrival, a direct gain, to apply the direct gain on the direct component signal to obtain a processed direct signal, and to combine the processed direct signal and a processed diffuse signal to generate the audio output signal. 1. A system for generating two or more audio output signals , comprising:a decomposition module,a signal processor, andan output interface,wherein the decomposition module is configured to receive two or more audio input signals, wherein the decomposition module is configured to generate a direct component signal, comprising direct signal components of the two or more audio input signals, and wherein the decomposition module is configured to generate a diffuse component signal, comprising diffuse signal components of the two or more audio input signals,wherein the signal processor is configured to receive the direct component signal, the diffuse component signal and direction information, said direction information depending on a direction of arrival of the direct signal components of the two or more audio input signals,wherein the signal processor is configured to generate one or more processed diffuse signals depending on the diffuse component signal,wherein, for each audio output signal of the two or more audio output signals, ...

SYSTEM, APPARATUS AND METHOD FOR CONSISTENT ACOUSTIC SCENE REPRODUCTION BASED ON ADAPTIVE FUNCTIONS

A system for generating one or more audio output signals is provided, having a decomposition module, a signal processor, and an output interface. The signal processor is configured to receive the direct component signal, the diffuse component signal and direction information. Moreover, the signal processor is configured to generate one or more processed diffuse signals depending on the diffuse component signal. For each audio output signal of the one or more audio output signals, the signal processor is configured to determine a direct gain, the signal processor is configured to apply the direct gain on the direct component signal to obtain a processed direct signal, and the signal processor is configured to combine the processed direct signal and one of the one or more processed diffuse signals to generate the audio output signal. The signal processor further has a gain function computation module and a signal modifier. 1. An apparatus for generating one or more audio output signals , comprising:a signal processor, andan output interface,wherein the signal processor is configured to receive a direct component signal, comprising direct signal components of two or more original audio signals, wherein the signal processor is configured to receive a diffuse component signal, comprising diffuse signal components of the two or more original audio signals, and wherein the signal processor is configured to receive direction information, said direction information depending on a direction of arrival of the direct signal components of the two or more original audio signals,wherein the signal processor is configured to generate one or more processed diffuse signals depending on the diffuse component signal,wherein, for each audio output signal of the one or more audio output signals, the signal processor is configured to determine, depending on the direction of arrival, a direct gain being a gain value, the signal processor is configured to apply said direct gain on the ...

SOUND PROCESSING APPARATUS AND SOUND PROCESSING SYSTEM

The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. 18-. (canceled)9. A method executed by at least one processing circuit of a sound processing apparatus , the method comprising:acquiring an audio object signal and metadata of the audio object signal, wherein the metadata is position data of an audio object associated with the audio object signal;{'claim-text': 'calculating the gain coefficients comprises calculating the gain coefficients for the plurality of sound outputting units based on a positional relationship among the plurality of sound outputting units and the sound image localization position; and', '#text': 'determining output gain coefficients for a plurality of sound outputting units located close to a sound image localization position as a target position at least in part by calculating gain coefficients for the plurality of sound outputting units based on a positional relationship between the plurality of sound outputting units with respect to each combination of a plurality of combinations of two or three of the plurality of sound outputting units, wherein:'}outputting the output gain coefficients for the plurality of sound outputting units.10. The method of claim 9 , wherein a number of the plurality of sound outputting units is equal to or larger than four.11. The method of claim 9 , wherein calculating the gain coefficients comprises calculating the gain coefficients for the plurality of sound outputting units based on position information of the plurality of sound outputting units.12. The method of claim 11 , wherein the position information of the plurality of sound outputting units is represented in a coordinate system.13. The method of claim 12 , wherein the coordinate system includes at least a vertical and horizontal direction.14. The method of claim 12 , wherein the coordinate system is a Cartesian coordinate system.15. The method of claim 12 , ...

APPARATUS AND METHOD FOR REPRODUCING MULTI-SOUND CHANNEL CONTENTS USING DLNA IN MOBILE TERMINAL

An acoustic apparatus and a method thereof in a mobile terminal are provided. More particularly, an apparatus and a method for establishing a multi-sound system that provides sounds of sense of reality by connecting with players existing in the neighborhood and reproducing channels of contents using a plurality of players when reproducing the contents including multi-sound channels are provided. The apparatus includes a control player, a server player, and a client player. The control player selects players that reproduce respective channels using information of the channels forming the multi-sound channel contents. The server player analyzes the contents to be reproduced by the players to provide the information regarding the channels to the control player. The client player reproduces the channels of the relevant contents under control of the control player. 114-. (canceled)15. An electronic device for reproducing multi-sound channel contents , the device comprising:a communication unit; and select a content comprising a plurality of channels among a plurality of contents,', 'determine channel information regarding the channels comprised in the selected content, and', 'select at least one second device to reproduce each channel comprised in the selected content using the determined channel information., 'a controller configured to16. The device of claim 15 , wherein the controller is configured to control to determine each channel information regarding the channel to be reproduced at each selected second device claim 15 , and transmit the each channel information to the each selected second device to allow the each selected second device to reproduce the channel corresponding to the received channel information.17. The device of claim 15 , wherein the controller is configured to control the selected at least one second device to reproduce the each channel of the content in response to a contents reproduction request.18. The device of claim 15 , wherein the ...

HEAD MOUNTED DISPLAY AND METHOD FOR PROVIDING AUDIO CONTENT BY USING SAME

The present invention relates to a head mounted display (HMD) for adaptively augmenting virtual audio signals according to an actual audio signal-listening environment, and to a method for providing audio content by using same. To this end, the present invention provides the HMD comprising: a processor for controlling the operation of the HMD; a microphone unit for receiving real sound; and an audio output unit for outputting a sound based on a command from the processor, wherein the processor receives the real sound using the microphone unit, obtains a virtual audio signal, extracts spatial audio parameters by using the received real sound; filters the virtual audio signal using the extracted spatial audio parameters, and outputs the filtered virtual audio signal to the audio output unit. 1. A method of providing audio contents by using a head mounted display (HMD) apparatus , the method comprising:receiving real sound by using a microphone;obtaining a virtual audio signal;extracting spatial audio parameters based on the received real sound;filtering the virtual audio signal by using the extracted spatial audio parameters; andoutputting the filtered virtual audio signal.2. The method according to claim 1 , wherein the spatial audio parameters include at least one of a reverberation time and a room impulse response (RIR) extracted by using the received real sound.3. The method according to claim 1 , wherein the virtual audio signal is obtained based on the received real sound.4. The method according to claim 1 , wherein claim 1 , in the outputting the virtual audio signal claim 1 , reproducing characteristics of the virtual audio signal are adjusted by using the received real sound.5. The method according to claim 4 , wherein the reproducing characteristics include at least one of a play pitch and a play tempo.6. The method according to claim 1 ,further comprising obtaining a position of a virtual sound source of the virtual audio signal,wherein, in the outputting ...

Method of improving externalization of virtual surround sound

Aspects of the present disclosure relate to techniques for processing a source audio signal in order to localize sounds. In particular, aspects of the present disclosure relate to sound localization techniques which externalize sounds for headphone audio, such as a virtual surround sound headphone system. In various implementations, room reverberations and other acoustic effects of the environment may be more accurately modeled using improved room reverberation models. For example, in some implementations, the underlying source signal may be filtered with a filter representing a room impulse response that is a combination of a stereo room impulse response and a mono room impulse response. By way of further example, in some implementations the source signal may be filtered with a combined impulse response filter that is derived from binaural recordings of simulated impulses recorded in a desired reverberant environment.

SOUND EFFECT CONTROLLING METHOD AND SOUND OUTPUTTING DEVICE WITH ORTHOGONAL BASE CORRECTION

A sound effect controlling method and a sound outputting device with orthogonal base correction are provided. The sound effect controlling method includes the following steps. Firstly, 6 or 8 initial sound signals are generated according to an original left sound signal and an original right sound signal. Next, a first gain is calculated according to a rotation angle and a first axial angle, and a second gain is calculated according to the rotation angle and a second axial angle, wherein the first axial angle and the second axial angle are orthogonal. Lastly, a modified left sound signal and a modified right sound signal are obtained according to the first gain and the second gain. 1. A sound effect controlling method with orthogonal base correction , comprising:receiving an original left sound signal and an original right sound signal;generating 6 or 8 initial sound signals according to the original left sound signal and the original right sound signal;detecting a rotation angle of a user;calculating a first gain according to the rotation angle and a first axial angle and calculating a second gain according to the rotation angle and a second axial angle, wherein the first axial angle and the second axial angle are orthogonal;transforming the initial sound signals to a first updated left sound signal and a first updated right sound signal by using the first axial angle as a center;transforming the initial sound signals to a second updated left sound signal and a second updated right sound signal by using the second axial angle as a center; andsynthesizing the first updated left sound signal and the second updated left sound signal as a modified left sound signal according to the first gain and the second gain and further synthesizing the first updated right sound signal and the second updated right sound signal as a modified right sound signal.2. The sound effect controlling method with orthogonal base correction according to claim 1 , wherein the first axial angle ...

REVERBERATION GENERATION FOR HEADPHONE VIRTUALIZATION

The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well. 1. A method of generating left-ear and right-ear binaural signals , the method comprising:determining a sound source location corresponding to each of one or more audio input signals;convolving each of said one or more audio input signals with one or more components of a BRIR corresponding to the sound source location to obtain left-ear and right-ear intermediate signals, wherein at least one of said components of the BRIR comprises directionally-controlled reflections that impart a particular perceptual cue to said one or more audio input signals respectively, the particular perceptual cure being selected from a plurality of perceptual cues, wherein the directionally controlled reflections are generated using a directional pattern which describes how directions of arrival of the directionally-controlled reflections change in relation to a direction of the sound source location as a function of time; andcombining the left-ear intermediate signals to produce the left-ear binaural signal and combining the right-ear intermediate signals to produce the right-ear binaural signal.2. The method of claim 1 , wherein the directional pattern includes a predetermined directional pattern having a wobble shape.3. The method of claim 1 , comprising providing the left-ear binaural signal and the right-ear binaural signal for headphone presentation.4. A system comprising:one or more ...

ORIENTATION-AWARE SURROUND SOUND PLAYBACK

Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed. 1. A method comprising:receiving, by an audio rendering system, one or more audio streams;generating one or more rendering components by the audio rendering system, the one or more rendering components including a rendering matrix R;applying different orientation compensations for direct and diffuse parts, respectively, of the rendering matrix R;determining an orientation dependent component O of the rendering matrix R, the orientation dependent component O being a function of an orientation in a three dimensional space;updating the orientation dependent component O according to an orientation of one or more electronic devices, the orientation of the one or more electronic devices being determined by one or more orientation sensors; anddispatching the one or more audio streams by the audio rendering system to one or more downstream devices according to the one or more rendering components including the orientation dependent component.2. The method of claim 1 , wherein the rendering matrix R includes an orientation independent component P.3. The method of claim 1 , wherein the orientation in a three-dimensional space is ...

Audio System Height Channel Up-Mixing

Audio system height channel up-mixing that is configured to develop two or more height channels from audio sources that do not include height-related encoding. The up-mixing involves determining correlations and normalized channel energies between input audio signals. At least two height channels (e.g., left and right height audio signals) are developed from the correlations and normalized energies.

SYSTEMS AND METHODS FOR PROVIDING AUGMENTED AUDIO

A system for providing augmented spatialized audio in a vehicle, including a plurality of speakers disposed in a perimeter of a cabin of the vehicle; and a controller configured to receive a position signal indicative of the position of a first user's head in the vehicle and to output to a first binaural device, according to the first position signal, a first spatial audio signal, such that the first binaural device produces a first spatial acoustic signal perceived by the first user as originating from a first virtual source location within the vehicle cabin, wherein the first spatial audio signal comprises at least an upper range of a first content signal, wherein the controller is further configured to drive the plurality of speakers with a driving signal such that a first bass content of the first content signal is produced in the vehicle cabin.

SOUND-REPRODUCING METHOD AND SOUND-REPRODUCING APPARATUS

A sound-reproducing method that includes the steps outlined below is provided. A playback sound is generated by applying original audio into a test environment. The playback sound is received to generate received sound data. At least one test environment spatial parameter corresponding to the test environment is calculated according to known audio data related to the original audio and the received sound data. Input audio is modified by applying the test environment spatial parameter thereto to generate reproduced audio. 1. A sound-reproducing method , comprising:generating a playback sound by applying original audio into a test environment, wherein the test environment is a virtual environment;receiving the playback sound to generate received sound data;calculating at least one test environment spatial parameter corresponding to the test environment according to the received sound data and known audio data related to the original audio; andmodifying input audio by applying the test environment spatial parameter thereto to generate reproduced audio.2. (canceled)3. (canceled)4. The sound-reproducing method of claim 1 , wherein the step of generating the playback sound further comprises:superimposing the original audio on an artificial audio corresponding to the test environment to generate the playback sound.5. The sound-reproducing method of claim 4 , further comprising:subtracting the received sound data and the known audio data to generate difference output audio data, wherein the difference output audio data comprises the at least one test environment spatial parameter.6. The sound-reproducing method of claim 1 , further comprising storing the test environment spatial parameter.7. The sound-reproducing method of claim 1 , wherein the test environment spatial parameter comprises a phase claim 1 , a time difference between channels claim 1 , a frequency response claim 1 , an amplitude or a combination thereof related to the received sound data and the known audio ...

Systems, Methods, and Apparatus for Recording Three-Dimensional Audio and Associated Data

Embodiments of the disclosure can include systems, methods, and apparatus for recording three-dimensional audio and associated data. In one embodiment, an apparatus can include a base, a stick operable to move in either an X-axis direction or a Y-axis direction with respect to the base, a pedal operable to mount to a portion of the stick and receive a user input in either the X-axis direction or the Y-axis direction, and a sliding mechanism operable to facilitate sliding the pedal with respect to the base, wherein the pedal is further operable to receive another user input to correspond to a Z-axis direction. In certain embodiments, the apparatus can also include a locking mechanism operable to receive a user input to limit sliding the pedal with respect to the base. 1. A system comprising:at least one processor in communication with at least one memory, the at least one processor operable to execute computer-executable instructions; receive a user input corresponding with sonic spatial data in a predefined listening environment; and', 'transmit the user input to at least one processor; and, 'an input device operable to receive the user input;', 'convert the user input to sonic spatial data;', 'associate the sonic spatial data with a time code; and', 'store the sonic spatial data, audio signal, and time code., 'the computer-executable instructions operable to2. The system of claim 1 , wherein the computer-executable instructions are further operable to:format the sonic spatial data, audio signal, and time code for storage to a 3D-EA audio output file.3. The system of claim 1 , wherein the input device comprises at least one of the following: a foot pedal claim 1 , a joystick device claim 1 , a hand controlled device claim 1 , light sensor claim 1 , a gesture control device claim 1 , or a voice input device.4. The system of claim 1 , wherein the user input comprises X claim 1 , Y claim 1 , and Z-axis coordinate data.5. The system of claim 1 , wherein the computer- ...

Systems, Methods, and Apparatus for Playback of Three-Dimensional Audio

Embodiments of the disclosure can include systems, methods, and apparatus for playback of three-dimensional audio sound. In one embodiment, a system can include at least one input module operable to receive a mono audio signal. The system can also include at least one processing module operable to convert the mono audio signal to a six channel signal, wherein the six channel signal is played back over a 3D-EA array of speakers, wherein the 3D-EA array of speakers facilitates a dome-shaped listening environment around a listener. In at least one embodiment, the at least one processing module is further operable to receive sonic spatial data, and based at least in part on the sonic spatial data, modify volume and/or delay associated with the six channel signal.

Digital audio filters for variable sample rates

Various exemplary embodiments relate to a method and apparatus for processing audio signals to influence the reproduction of the audio signals. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive a virtualization profile including a digital audio filter with a design sample rate, resample the virtualization profile to a different sample rate, filter the audio signal with the resampled virtualization profile, and reproduce the filtered audio signal as sound.

ENHANCED MULTICHANNEL AUDIO INTERCEPTION AND REDIRECTION FOR MULTIMEDIA DEVICES

A system and method for enhanced multichannel audio interception and redirection for ANDROID™-based devices, comprising an audio redirector software module that detects available hardware capabilities, configures and reports audio channel capabilities based on the detected hardware capabilities, up-mixes source audio to match available output channels on destination devices, and provides up-mixed audio channels to both a sound processing framework and external audio rendering hardware. 1. A system for multichannel audio interception and redirection for multimedia devices , comprising: receive an audio stream from an audio source;', 'determine a number of audio channels playable by the audio output of the mobile device;', 'perform a de-multiplexing operation on the audio stream to produce a plurality of audio channels;', 'send a first subset of the de-multiplexed audio channels, comprising the determined number of audio channels, to the audio output of the mobile device; and', 'send a second subset of the de-multiplexed audio channels to a plurality of external audio playback devices via a wireless network., 'a mobile computing device comprising at least a processor, a memory, an audio output, and a plurality of programming instructions stored in the memory and operable on the processor, wherein the plurality of programming instructions, when operating on the processor, cause the processor to2. The system of claim 1 , further comprising a packet concentrator configured to receive outbound audio channel data and to modify the outbound audio channel data claim 1 , the modification comprising at least modifying the size of at least a portion of the packets in the audio channel data claim 1 , and configured to transmit the modified audio channel data to the plurality of external hardware devices via a network.3. The system of claim 2 , wherein the modifications comprise embedding synchronization event data into the outbound audio channel data claim 2 , the ...

BASS MANAGEMENT FOR HOME THEATER SPEAKER SYSTEM AND HUB

An audio system is disclosed that contains speakers that are configured to transmit unique parameters to an audio source. The system includes a hub that is configured to be connected with one or more audio sources. At least one speaker is included having one or more filter parameters stored internally that are specific to the speaker. The speaker is operable to transmit the one or more filter parameters to the hub. The hub is operable to use the one or more filter parameters to filter an audio channel associated with the speaker and an associated subwoofer channel. 1. A system , comprising:a hub configured to be connected with one or more audio sources; andat least one speaker having one or more filter parameters stored internally that are specific to said at least one speaker, wherein said at least one speaker is operable to transmit said one or more filter parameters to said hub, and wherein said hub is operable to use said one or more filter parameters to filter an audio channel associated with said at least one speaker and an associated subwoofer channel.2. The system of claim 1 , wherein said one or more filter parameters is transmitted to said hub over a wireless communication link.3. The system of claim 1 , wherein said one or more filter parameters is transmitted to said hub over a wired communication link.4. The system of claim 1 , wherein said at least one speaker includes at least one amplifier.5. The system of claim 1 , wherein said hub has at least two filters for processing a first audio signal for said at least one speaker and at least two filters for processing a second audio signal for said subwoofer channel.6. The system of claim 1 , wherein said hub implements said one or more filter parameters as one or more crossover filters.7. The system of claim 6 , wherein said one or more crossover filters are implemented as finite impulse response (FIR) filters.8. The system of claim 6 , wherein said one or more crossover filters are implemented as infinite ...

HEADSET DEVICE AND A DEVICE PROFILE MANAGEMENT SYSTEM AND METHOD THEREOF

Most computer operating systems are able to automatically configure the coupled computer peripheral audio device for use with the computer operating system without the need for installation of a specific driver. However, when these computer peripheral audio devices are detected by the computing system, a generic audio device control configuration is often assigned to them and whatever customised settings previously configured by the user will be lost and replaced with a new set of unfamiliar setting. This poses much inconvenience and hassle to gamers when they use a different computing system or machine. Described according to an embodiment of the invention is a headset device and a device profile management method, operating on a computing cloud, comprising steps for receiving client data provided by a computing system detecting coupling of an audio device thereto and for retrieving configuration data corresponding to the client data for use in configuring the audio device by the computing system. 1. A profile management method comprising:receiving client data provided by a computing system, the client data comprising device data being descriptive of an audio device in signal communication with the computing system, the device data being further descriptive of the user interactions identifiable by the audio device;associating one of a plurality of device profiles with the device data, each of the plurality of device profiles having configuration data associated therewith; andproviding the configuration data associated with the one of the plurality of device profiles to the computing system for configuring operation of the audio device with the computing system in response to at least a portion of the user interactions identifiable by the audio device;wherein the audio device is a computer peripheral audio device coupled with an interface formed integral with the computing system.2. (canceled)3. The profile management method as in claim 1 , wherein the audio device ...

SIMULATING ACOUSTIC OUTPUT AT A LOCATION CORRESPONDING TO SOURCE POSITION DATA

Systems and methods of simulating acoustic output at a location corresponding to source position data are disclosed. A particular method includes receiving an audio signal and source position data associated with the audio signal. A set of speaker signals are applied to a plurality of speakers, where the set of speaker driver signals causes the plurality of speakers to generate acoustic output that simulates output of the audio signal by an audio source at a location corresponding to the source position data. 1. A method comprising:receiving an audio signal and source position data associated with the audio signal; andapplying a set of speaker driver signals to a plurality of speakers, wherein the set of speaker driver signals causes the plurality of speakers to generate acoustic output that simulates output of the audio signal by an audio source at a location corresponding to the source position data.2. The method of claim 1 , wherein the set of speaker driver signals corresponds to one or more fixed speakers claim 1 , one or more virtual speakers claim 1 , or a combination thereof.3. The method of claim 1 , wherein the location corresponding to the source position data is distinct from locations of the plurality of speakers.4. The method of claim 1 , further comprising applying a second set of speaker driver signals to the plurality of speakers to generate acoustic output corresponding to a second location that is different from the location.5. The method of claim 1 , wherein the audio signal claim 1 , the source position data claim 1 , or both are received from an automatic driver assistance system claim 1 , a navigation system claim 1 , or a mobile device.6. The method of claim 1 , further comprising:up-mixing the audio signal to generate a plurality of intermediate signal components;down-mixing the plurality of intermediate signal components to generate a plurality of speaker signal components; andprocessing the plurality of speaker signals components to ...

Method and System for Surround Sound Processing in a Headset

An audio headset may receive a plurality of audio signals corresponding to plurality of surround sound channels. The headset may determine, via its audio processing circuitry, context and/or content of the audio signals. The audio processing circuitry may process the audio signals to generate stereo signals carrying one or more virtual surround channels, wherein the processing comprises automatically controlling, based on the context and the content of the audio signals, a simulated acoustic environment of the virtual surround channels. 1. A method , comprising:in an audio headset that receives a plurality of rendered surround sound channels:determining, via audio processing circuitry, a type of audio carried by said surround sound channels; andprocessing, in said audio processing circuitry, said surround sound channels to generate stereo signals carrying one or more virtual surround channels, wherein said processing comprises controlling, based on said type of audio carried by said surround sound channels, a simulated acoustic environment of the virtual surround channels.2. The method of claim 1 , wherein:said determining said type of audio comprises distinguishing between game audio, music audio, and movie audio.3. The method of claim 2 , comprising:when said type of audio is game audio, automatically selecting, by said audio processing circuitry, a first simulated acoustic environment;when said type of audio is music audio, automatically selecting, by said audio processing circuitry, a second simulated acoustic environment; andwhen said type of audio is movie audio, automatically selecting, by said audio processing circuitry, a third simulated acoustic environment.4. The method of claim 2 , wherein:when said type of said audio carried by said surround sound channels is music audio, said processing said surround sound channels to generate said stereo signals comprises attenuating, by said audio processing circuitry, side audio channels and rear audio channels of ...

SYSTEM AND METHOD FOR PERFORMING PANNING FOR AN ARBITRARY LOUDSPEAKER SETUP

Placement of one or two placed virtual loudspeakers within a loudspeaker setup that includes a real loudspeakers is determined and vector base amplitude panning (VBAP) gains including the gains of the real loudspeakers and placed one or two virtual loudspeakers are also then determined. Gains of one or two placed virtual loudspeakers are redistributed to the real loudspeakers to ensure preservation of total energy. Real loudspeakers in the loudspeaker setup have redistributed gains of one or two placed virtual loudspeakers. Loudspeaker outputs are generated and transmitted to the real loudspeakers to be played back. When received audio content is ambisonics content, a predetermined grid is generated and HOA content is projected to the grid. Other aspects are also described. 1. A method of performing panning for an arbitrary loudspeaker setup , comprising: a) when the loudspeaker setup is a 2-channel setup, determining locations of the two virtual loudspeakers based on a center of a line formed by locations of two real loudspeakers in the loudspeaker setup and a listening position,', 'b) when the loudspeaker setup is a 2-dimensional (2D) setup that refers to more than two real loudspeakers, determining the locations of the two virtual loudspeakers based on a centroid of a polygon formed by locations of real loudspeakers and the listening position, or', 'c) when the loudspeaker setup is a 3-dimensional (3D) setup, determining the location of the one virtual loudspeaker based on a center of gravity of a polyhedron formed by the positions of the real loudspeakers; and, 'determining a placement of one or two virtual loudspeakers within a loudspeaker setup, wherein the loudspeaker setup also refers to a plurality of real loudspeakers, wherein the determining comprises one of'}determining vector base amplitude panning (VBAP) gains, wherein the VBAP gains include gains of the real loudspeakers and of the one or two virtual loudspeakers; andgenerating loudspeaker outputs for ...

Manipulation of Playback Device Response Using Signal Processing

Example techniques involve balancing bass response of a multiple driver playback device. An example playback device receives left and right channels of audio content and generates a center channel of the audio content by combining at least a portion of the left right channels. The playback device generates first and second side channels of the audio content by combining the center channel and a difference of the left channel and the right channel and combining the center channel and an inverse of the difference of the left channel and the right channel, respectively. The first and second side channels are attenuated by a filter with a given cutoff frequency. The center channel is amplified in proportion to the attenuation of the first side channel. The playback device plays back the first side channel, second side channel, and center channel according to respective radiation patterns having maximums aligned in respective directions. 1. A method of operating a playback device , the method comprising:receiving, at the playback device, left and right channels of audio content;generating a center channel of the audio content, wherein generating the center channel comprises combining at least a portion of the left channel and at least a portion of the right channel; wherein generating the first side channel comprises combining (i) the center channel and (ii) a difference of the left channel and the right channel, and', 'wherein generating the second side channel comprises combining (i) the center channel and (ii) an inverse of the difference of the left channel and the right channel;, 'generating first and second side channels of the audio content,'} applying one or more filters to attenuate portions of the first side channel having frequencies less than a cutoff frequency;', 'playing back the first side channel according to a first radiation pattern having a maximum aligned with a first direction;', 'playing back the second side channel according to a second radiation ...

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 correctly using the available speaker positions in a room that is set up to play the adaptive audio content. 1. A system for processing audio signals , comprising an authoring component configured to:receive a plurality of audio signals;generate an adaptive audio mix comprising a plurality of monophonic audio streams and metadata associated with each of the audio streams and indicating a playback location of a respective monophonic audio stream, wherein at least some of the plurality of monophonic audio streams are identified as channel-based audio and the others of the plurality of monophonic audio streams are identified as object-based audio, and wherein the playback location of a channel-based monophonic audio stream comprises a designation of a speaker in a speaker array, and the playback location of an object-based monophonic audio stream comprises a location in three-dimensional space, and wherein each object-based monophonic audio stream is rendered in at least one specific ...

CODING OF MULTICHANNEL AUDIO CONTENT

There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration. 1receiving a plurality of input audio signals, the plurality of input audio signals including a first waveform-coded signalcomprising spectral data corresponding to frequencies up to a first frequency anda second waveform-coded signal comprising spectral data corresponding to frequencies up to a second frequency, the second frequency being higher than the first frequency;decoding the first waveform-coded signal to produce a first decoded audio signal having frequencies up to the first frequency, the first decoded audio signal representing a side signal;decoding the second waveform-coded signal to produce a second decoded audio signal having frequencies up to the second frequency, the second decoded audio signal representing a mid signal;performing an enhanced inverse sum-difference transformation with the first decoded signal and the second decoded signal to produce a stereo audio signal up to the first frequency, wherein the enhanced inverse sum-difference transformation includes applying a weighting parameter to the mid signal;performing an inverse sum-difference transformation with the second decoded signal to produce a stereo audio signal up to the second frequency; andcombining the stereo audio signal having frequencies up to the first frequency with ...