ДЕКОДЕР И СПОСОБ ПАРАМЕТРИЧЕСКОЙ КОНЦЕПЦИИ ОБОБЩЕННОГО ПРОСТРАНСТВЕННОГО КОДИРОВАНИЯ АУДИООБЪЕКТОВ ДЛЯ СЛУЧАЕВ МНОГОКАНАЛЬНОГО ПОНИЖАЮЩЕГО МИКШИРОВАНИЯ/ПОВЫШАЮЩЕГО МИКШИРОВАНИЯ

Изобретение относится к средствам для пространственного кодирования аудиообъектов. Технический результат заключается в повышении качества кодирования аудиообъектов. Декодер для генерирования выходного аудиосигнала содержит один или более выходных аудиоканалов, из микшированного с понижением сигнала, содержащего один или более каналов понижающего микширования. Микшированный с понижением сигнал кодирует один или более сигналов аудиообъектов. Декодер содержит определитель порога для определения порогового значения в зависимости от энергии сигнала и/или энергии шума, по меньшей мере, одного из упомянутых одного или более сигналов аудиообъектов и/или в зависимости от энергии сигнала и/или энергии шума, по меньшей мере, одного из упомянутых одного или более каналов понижающего микширования. Более того, декодер содержит блок обработки для генерирования упомянутых одного или более выходных аудиоканалов из упомянутых одного или более каналов понижающего микширования в зависимости от порогового значения ...

ДИСПЛЕЙ И УСТРОЙСТВО ВЫВОДА ЗВУКА

Изобретение относится к области видео/аудио воспроизведения. Технический результат - сохранение качества звука, который вызывает высокое ощущение реальности. Дисплей, содержащий: часть дисплея, на которой отображают видеоизображение; первую часть вывода звука, которая выводит стереофонический звук в высокочастотном диапазоне, и которая представляет собой поверхностный источник звука, расположенный на задней поверхности части дисплея, на одном из верхнего участка и нижнего участка части дисплея; вторую часть вывода звука, которая выводит стереофонический звук в более низком частотном диапазоне, чем первая часть вывода звука и которая представляет собой один из поверхностного источника звука и точечного источника звука, расположенного на задней поверхности части дисплея, на одном из верхнего участка и нижнего участка части дисплея, на котором не расположена первая часть вывода звука; часть задержки, которая выполняет задержку вывода второй части вывода звука для ее вывода с запаздыванием ...

УСТРОЙСТВО ОБРАБОТКИ АУДИОСИГНАЛА И СПОСОБ ФИЛЬТРАЦИИ АУДИОСИГНАЛА

Изобретение относится к средствам для обработки аудиосигнала. Технический результат заключается в обеспечении объемного звука без перекрёстных помех. Определяют матрицу фильтра на основании матрицы акустической передаточной функции и матрицы целевой акустической передаточной функции. Матрица целевой акустической передаточной функции содержит целевые передаточные функции целевых путей распространения акустического сигнала. При этом целевые пути распространения акустического сигнала определяются целевой конфигурацией множества положений виртуальных громкоговорителей относительно слушателя. Фильтруют входной аудиосигнал левого канала на основании матрицы фильтра для получения первого фильтрованного входного аудиосигнала левого канала и второго фильтрованного входного аудиосигнала левого канала. Фильтруют входной аудиосигнал правого канала на основании матрицы фильтра для получения первого фильтрованного входного аудиосигнала правого канала и второго фильтрованного входного аудиосигнала правого ...

СПОСОБ И УСТРОЙСТВО ДЛЯ ВОСПРОИЗВЕДЕНИЯ ТРЕХМЕРНОГО ЗВУКОВОГО СОПРОВОЖДЕНИЯ

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

УСТРОЙСТВО И СПОСОБ ОБЕСПЕЧЕНИЯ МЕРЫ ПРОСТРАНСТВЕННОСТИ, СВЯЗАННОЙ С АУДИОПОТОКОМ

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

ПРОСТРАНСТВЕННАЯ ЗВУКОВОСПРОИЗВОДЯЩАЯ СИСТЕМА

Пространственная звуковоспроизводящая система относится к радиотехнике и может быть использована для озвучивания различных помещений при индивидуальном прослушивании. Система содержит многоканальный источник сигнала и каналы звуковоспроизведения. Каждый канал звуковоспроизведения выполнен в виде усилителя низкой частоты и громкоговорителя, связанных последовательно, зондирующего устройства, устройства автоматического регулирования, линии связи, при этом выход зондирующего устройства посредством линии связи подключен к первому входу устройства автоматического регулирования, второй вход которого подключен к выходу источника сигнала, а выход связан с входом усилителя низкой частоты. 18 з.п.ф-лы, 4 ил.

ДИСПЛЕЙ И УСТРОЙСТВО ВЫВОДА ЗВУКА

... 1. Дисплей, содержащий: ! часть дисплея, на которой отображают видеоизображение; !первую часть вывода звука, которая выводит стереофонический звук в высокочастотном диапазоне, и которая представляет собой поверхностный источник звука, расположенный на задней поверхности части дисплея, на одном из верхнего участка и нижнего участка части дисплея; ! вторую часть вывода звука, которая выводит стереофонический звук в более низком частотном диапазоне, чем первая часть вывода звука, и которая представляет собой один из поверхностного источника звука и точечного источника звука, расположенного на задней поверхности части дисплея, на одном из верхнего участка и нижнего участка части дисплея, на котором не расположена первая часть вывода звука; и ! часть задержки, которая выполняет задержку вывода второй части вывода звука для ее вывода с запаздыванием относительно вывода первой части вывода звука. ! 2. Дисплей по п.1, !в котором область наложения формируют между частотами вывода звука первой части ...

СТЕРЕОФОНИЧЕСКОЕ РАСШИРЕНИЕ

... 1. Способ, содержащий этапы, на которых: ! - осуществляют доступ входного стереосигнала в систему воспроизведения звука, которая включает в себя, по меньшей мере, два громкоговорителя; ! - при этом стереосигнал включает в себя множество частотных компонентов; и ! - при этом, по меньшей мере, два громкоговорителя располагаются в пространственной близости друг к другу; ! - декоррелируют высокочастотный диапазон частотных компонентов, при этом декоррелированный высокочастотный диапазон соответствует высоким частотам выше пороговой частоты, при этом упомянутая пороговая частота находится между 300 Гц и 3 кГц при отсутствии декорреляции диапазона более низких частот; и ! - расширяют стереофоническую характеристику системы воспроизведения звука на основе этапа декорреляции. ! 2. Способ по п.1, дополнительно содержащий этап, на котором: ! - предварительно обрабатывают стереосигнал; ! - при этом этап предварительной обработки включает в себя этап декорреляции. ! 3. Способ по п.1, в котором близость ...

СПОСОБ И УСТРОЙСТВО ДЛЯ РАЗЛОЖЕНИЯ СТЕРЕОФОНИЧЕСКОЙ ЗАПИСИ С ИСПОЛЬЗОВАНИЕМ ОБРАБОТКИ В ЧАСТОТНОЙ ОБЛАСТИ, ПРИМЕНЯЮЩЕЙ ГЕНЕРАТОР СПЕКТРАЛЬНЫХ ВЕСОВ

... 1. Устройство для формирования стереофонического бокового сигнала, содержащего первый боковой канал и второй боковой канал, из стереофонического входного сигнала, содержащего первый входной канал и второй входной канал, содержащее:генератор (110; 610; 710; 810; 910; 1310) информации об изменении для формирования информации об изменении, причем генератор (110; 610; 710; 810; 910; 1310) информации об изменении содержит генератор (116; 615; 715; 815; 915) спектральных весов для формирования информации об изменении путем формирования первого спектрального весового коэффициента на основании монофонического среднего сигнала и на основании монофонического бокового сигнала в стереофоническом входном сигнале, иблок (120; 620; 720; 820; 1320) манипулирования сигналами, выполненный с возможностью манипулирования первым входным каналом на основании информации об изменении для получения первого бокового канала, и выполненный с возможностью манипулирования вторым входным каналом на основании информации ...

CLAY/TONE SYSTEM AND PROCEDURE FOR THE TONWIEDERGABE

ELEKTROAKUSTISCHE ANORDNUNG ZUR DARBIETUNG VON AUDIOSIGNALEN FUER DAS RICHTUNGSORIENTIERTE RAEUMLICHE HOEREN BEI BINAURALER WIEDERGABE

An electroacoustic method of presenting audio signals for direction-oriented stereophonic listening by the binaural reproduction and transmission method is characterised in that to ensure clear perception of the direction of the sound event directed towards a listener, in addition to the primary signal produced a second and delayed signal is presented by a sound source (L2) disposed on a line connecting the same to the primary sound source (L1) and opposite the listener (H), the delay or lag of the second signal relative to the primary signal being greater than is required by sum localisation but remaining less than is required to hear the first echo. ...

AUDIO/CVIDEO SYSTEM AND PROCEDURE, A USER PERMITTING, DIFFERENT ONE, AN EVENT ASSIGNED OPINIONS AND SOUNDS AUSZUWÄHLERN

PRODUCTION OF DEKORRELIERTER SIGNALS

ELECTROACOUSTIC ARRANGEMENT FOR THE PRESENTATION OF AUDIO SIGNALS FOR THE DIRECTION-ORIENTED SPATIAL HEARING DURING BINAURALER RENDITION

VIDEODISC PLAYER FOR THE RENDITION OF NOTED SIGNALS

ELECTROACOUSTIC TRANSFORMATION WITH DEVICES FOR THE IMPROVEMENT OF THE LOW FREQUENCIES

CLAY/TONE OUTPUT SYSTEMS.

MULTI-PATH MORE AUDIODEKODER

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

Method and device for developing a virtual speaker distant from the sound source

Apparatus and method for encoding a spatial audio representation or apparatus and method for decoding an encoded audio signal using transport metadata and related computer programs

An apparatus for encoding a spatial audio representation representing an audio scene to obtain an encoded audio signal, comprises: a transport representation generator (600) for generating a transport representation (611) from the spatial audio representation, and for generating transport metadata (610) related to the generation of the transport representation (611) or indicating one or more directional properties of the transport representation (611); and an output interface (640) for generating the encoded audio signal, the encoded audio signal comprising information on the transport representation (611), and information on the transport metadata (610).

SOUND IMAGING METHOD AND APPARATUS

REVERBERATION APPARATUS

SOUND REPRODUCTION SYSTEM, PROGRAM AND DATA CARRIER

Method and system for head-related transfer function generation by linear mixing of head-related transfer functions

A method for performing linear mixing on coupled Head-related transfer functions (HRTFs) to determine an interpolated HRTF for any specified arrival direction in a range (e.g., a range spanning at least 60 degrees in a plane, or a full range of 360 degrees in a plane), where the coupled HRTFs have been predetermined to have properties such that linear mixing can be performed thereon (to generate interpolated HRTFs) without introducing significant comb filtering distortion. In some embodiments, the method includes steps of: in response to a signal indicative of a specified arrival direction, performing linear mixing on data indicative of coupled HRTFs of a coupled HRTF set to determine an HRTF for the specified arrival direction; and performing HRTF filtering on an audio input signal using the HRTF for the specified arrival direction.

Stereo to enhanced spatialisation in stereo sound hi-fi decoding process method and apparatus

Acoustic heterodyne device and method

STEREOPHONIC SOUND REPRODUCTION SYSTEM

In a stereophonic sound reproduction system in which stereophonic left and right signals are treated to have is level difference therebetween for establishing a localized sound image spaced from the midpoint between left and right loudspeakers, the level difference between such left and right signals is reduced as the frequency thereof increases for improving the localization of the sound image resulting from the application of the resulting or converted left and right signals to the left and right loudspeakers.

PARAMETRIC VIRTUAL SPEAKER AND SURROUND-SOUND SYSTEM

A system for generating at least one remote virtual speaker location in connection with at least a partial reflective environment (12, 12, 14 or 15) in combination with an audio speaker for creating multiple sound effects including a virtual sound source from the reflective environment which is perceived by a listener (53) as an original sound source, by generating a primary direct audio output by emitting audio compression waves toward a listener, and generating a secondary indirect audio output from at least one virtual speaker (24, 25 or 26) remote from the audio speakers, by emitting ultrasonic sound from at least one parametric speaker (20, 21 or 22) associated with the audio speakers and oriented toward at least one reflective environment which is remote from the audio speakers.

SOUND IMAGE CONTROL SYSTEM

To provide a sound image control system able to concurrently perform sound image localization control for two persons . The present invention is directed to a sound image control system controlling a sound image localization position by reproducing an audio signal from a plurality of loudspeakers . The sound image control system includes at least four loudspeakers for reproducing the audio signal and a signal processing section. The signal processing section sets four points corresponding to both ears of two listeners as control points , and performs signal processing for the audio signal input into the plurality of loudspeakers so that two target sound source positions indicating sound image localization positions of the respective two listeners are set in the same direction with respect to the two listeners . Here, the two target sound source positions are set so as to satisfy a following condition, T1 Подробнее

METHOD AND APPARATUS FOR DECOMPOSING A STEREO RECORDING USING FREQUENCY-DOMAIN PROCESSING EMPLOYING A SPECTRAL WEIGHTS GENERATOR

An apparatus for generating a stereo side signal having a first side channel and a second side channel from a stereo input signal having a first input channel and a second input channel is provided. The apparatus comprises a modification information generator (110) for generating modification information based on mid-side information. Furthermore, the apparatus comprises a signal manipulator (120) being adapted to manipulate the first input channel based on the modification information to obtain the first side channel and being adapted to manipulate the second input channel based on the modification information to obtain the second side channel. The modification information generator (110) comprises a spectral weights generator ( 116) for generating the modification information by generating a first spectral weighting factor based on a mono mid signal and on a mono side signal of the stereo input signal.

METHOD AND APPARATUS FOR REPRODUCING STEREOPHONIC SOUND

Method and apparatus reproduce a stereophonic sound. The method includes obtaining sound depth information which denotes a distance between at least one object within a sound signal and a reference position, and providing sound perspective to the sound object output from a speaker, based on the sound depth information.

Moving Picture Coding Apparatus

Passive Surround Sound Circuit

ACOUSTIC CORRECTION APPARATUS

An acoustic correction apparatus (20) processes a pair of left and right input signals to compensate for spatial distortion as a function of frequency when said input signals are reproduced through speakers in a sound system. The sound-energy of the left and right input signals (26 and 28) is separated and corrected in a first low-frequency range and a second high-frequency range. The resultant signals are recombined to create image-corrected audio signals (27 and 29) having a desired sound-pressure response when reproduced by the speakers in the sound system. The desired sound-pressure response creates an apparent sound image location with respect to a listener. The image-corrected signals (27 and 29) may then be spatially-enhanced to broaden the apparent sound image.

Verfahren zur Übertragung von stereophonischen Signalen.

Verfahren zur Aufzeichnung und Wiedergabe von Schalldarbietungen.

Verfahren zur zweikanaligen Aufnahme, Aufzeichnung, Übertragung und Wiedergabe von Schallereignissen

ACOUSTIC HETERODYNE DEVICE AND METHOD

Directional audio notification

Speaker system, audio amplifier and audio system

Process for the stereophomic reproduction of the sounds

ELECTROACOUSTIC METHOD FOR PRESENTING AUDIO SIGNALS FOR LISTENING STEREOPHONIC EFFECT STEERING

STEREOPHONIC INSTALLATION

DELAY ELEMENT

METHOD AND DEVICE FOR GENERATING POWER SIGNALS FOR A SOUND REPRODUCTION SYSTEM

NONRECURSIVE SYSTEM TO EXTEND the STEREOPHONIC BASE Of an APPARATUS OF ACOUSTIC DIFFUSION STEREOPHONIC

Computer system, has processing module having selection module which re-assembles stored audio signals into different groups, to determine representative spatial position data of signals

L'invention concerne un dispositif informatique comprenant une mémoire 108 pour stocker des signaux audio 114 en partie pré-enregistrés, correspondant chacun à une source définie par des données de position spatiale 116, un module de traitement 110 pour traiter ces signaux audio en temps réel en fonction des données de position spatiale. Le module de traitement 110 permet de calculer des paramètres de niveau de puissance instantané à partir des signaux audio 114, les source correspondantes étant définies par les paramètres de niveau de puissance instantané. Le module de traitement 110 comprend un module de sélection 120 pour regrouper certains des signaux audio en un nombre variable de groupes de signaux audio et le module de traitement 110 peut calculer des données de position spatiale représentatives d'un groupe de signaux audio en fonction des données de position spatiale 116 et des paramètres de niveaux de puissance instantanés de chaque source correspondante.

NEW METHOD OF LESSENING THE ACOUSTIC PHASE DISTORTIONS IN INSTALLATIONS FOR AUTOMOTIVE SOUND REPRODUCTION

Nouveau procédé de diminution des distorsions de phase acoustique dans les installations de reproduction sonore en automobile. Ce procédé utilise la différence entre les distorsions acoustiques aux places avant gauche et droite principalement dans les bandes de fréquence pour lesquelles elles ne sont pas de même signe.

Method and apparatus for synthesizing stereo signal

2채널 오디오에서의 스테레오 음장 개선을 위한 시스템 및 방법

... 본 발명은 2채널 오디오 시스템 및/또는 응용에서의 오디오 신호를 디지털 처리하는 방법 및 시스템을 제공한다. 구체적으로, 본 발명은 2채널 오디오 입력 신호를 낮은 주파수 신호와 이보다 높은 고역 주파수 신호로 분할하도록 구성된 제1 필터를 포함한다. M/S 분할기는 고역 주파수 신호를 중앙 신호와 측면 신호로 분할하도록 구성된다. 검파 모듈은 중앙 신호로부터 검파 신호를 생성하도록 설정되는데, 이 검파 신호는 측면 신호를 변조하도록 구성된 압축 모듈에서 게인변조된 측면 신호를 만드는 데 사용된다. 처리 모듈은 낮은 주파수 신호, 중앙 신호, 및 게인변조된 측면 신호를 결합하여서 최종 출력 신호를 형성하도록 구성된다.

BASS ENHANCEMENT SYSTEM

APARELHO E RECEPTOR DE TELEVISAO

Sub-band spatial audio enhancement

Converter and method for converting an audio signal

A converter and conversion method are disclosed for converting N channel audio input channels into M channel audio output channels, wherein a processor is used for applying a transfer function to a signal received on an input channel to obtain reverberation components of a calculated output channel, wherein said transfer function is a simplified transfer function matching a selected subset of a set of local maxima of a measured reverberation when applied to a corresponding stimulus.

Method and device for digital audio signal processing

A method and a device for digital audio signal processing are disclosed, wherein two speakers are used to simulate stereo sounds with planar effect. The simulation method is performed by designing a structure which is processed by a plurality of analog devices, such as filters, gain amplifiers, delay processors, and the like. The present invention also increases the effect of the SBS sounds and the stereo sounds thereby making the audio output of a digital home audio-video device more realistic.

SOUND SYSTEM AND METHOD OF SOUND REPRODUCTION

A sound reproduction system comprises a left and right speakers located in close proximity, and a sound processor which provides audio signals to the pair of speakers. The sound processor preferably derives a cancellation signal from the difference between the left and right channels. The resulting difference signal is scaled, delayed (if necessary), and spectrally modified before being added to the left channel and, in opposite polarity, to the right channel. The spectral modification to the difference channel preferably takes the form of a low-frequency boost over a specified frequency range, in order to restore the correct timbral balance after the differencing process. Additional phase-compensating all-pass networks may be inserted in the difference channel to correct for any extra phase shift contributed by the spectral modifying circuit. The technique may be used in a surround sound system or an automobile sound system. In an automobile, the pair of speakers may be placed together ...

METHODS AND APPARATUS FOR SUB-HARMONIC GENERATION, STEREO EXPANSION AND DISTORTION

Methods and apparatus are disclosed that achieve sub-harmonic signal processing (120, 134, 150), stereo-width expansion (400), sub-woofer signal processing, and tube distortion emulation (500) to achieve various desirable acoustic effects when used to modify an input signal containing, for example, music content.

GRAPHIC BALANCER

A graphic balancer having level adjusting means (operation amplifiers 1L, 1R, resonance circuits 2L, 2R, volumes 3L, 3R) for each of the bands to increase the level of voice signals of one channel between the stereophonic two channels divided into a plurality of bands, and to correspondingly decrease the level of the other channel, so that the position of acoustic image can be changed for each band. An embodiment of the invention is shown, in which the graphic balancer and the graphic equalizer can be switched to operate.

HYPERBOLIC PARABOLOID AUDIO SYSTEM

By combining a virtual sound system and a novel five-face directive speaker, the sound reproducing technique of the prior art is overcome. A stereophonic sound quality is created while upsetting the mechanical sound generation of the prior art and while stressing the hyperbolic paraboloidal processing by the angle of generation of sound. The sound waves generated from the five faces cross at the individual angles, differential signals are produced the frequency correction is made by unique sound flows, and sound is collected thereby to reproduce a sound field which is effective in the stereophonic real sound.

PARAMETRIC CODING OF SPATIAL AUDIO WITH OBJECT-BASED SIDE INFORMATION

A binaural cue coding scheme involving one or more object-based cue codes, wherein an object-based cue code directly represents a characteristic of an auditory scene corresponding to the audio channels, where the characteristic is independent of number and positions of loudspeakers used to create the auditory scene. Examples of object-based cue codes include the angle of an auditory event, the width of the auditory event, the degree of envelopment of the auditory scene, and the directionality of the auditory scene.

Dimensional sound recording and apparatus and method for producing the same

A sound recording having right and left information channels to drive right and left speakers, respectively. The right information channel has a left main stereo component having a pattern corresponding and similar to a left stereo input signal. In addition, it has a right to left compensating component having a pattern which corresponds to an inverted and delayed pattern of a right stereo input signal. In like manner, the right channel has a right main stereo component having a pattern corresponding and similar to the right stereo input signal. It also has a left to right compensating component having a pattern which corresponds to an inverted and delayed pattern of the left stereo input signal. The effect is that the main stereo components travelling to the nearest ear of a listener are not diminished substantially, while the main stereo components travelling to the further ear are diminished. The effect is a highly dimentionalized sound pattern.

Acoustic image enhancing method and apparatus

A method and apparatus are described for processing audio signals in a particular manner to achieve an enhancement in the perceived sound when the processed audio signals are projected by audio transducers. A predetermined low frequency band of the audio signals is delayed by a time period which is generally less than the refractory period of the human ear and combined with out-of-phase cross-field portions within a particular amplitude range. A predetermined low frequency boost is applied generally of the order of about 3 db. Several embodiments are described to provide improved recordings or enhanced acoustic projections of conventional stereo recordings.

Three-dimensional virtual audio display employing reduced complexity imaging filters

A three-dimensional virtual audio display method is described which includes generating a set of transfer function parameters in response to a spatial location or direction signal. An audio signal is filtered in response to the set of transfer function parameters. The set of transfer function parameters are selected from or interpolated among parameters derived by smoothing frequency components of a known transfer function over a bandwidth which is a non-constant function of frequency. The smoothing includes for each frequency component in at least part of the audio band of the display, applying a mean function to the amplitude of the frequency components within the bandwidth containing the frequency component, and noting the parameters of the resulting compressed transfer function.

Noise reducing circuit for CTD delay line

In a delay circuit for delaying an input signal having a range extending over a higher- and a lower-frequency region, and in which a charge transfer device, such a bucket-brigade device, transfers a signal from an input to an output thereof in response to a clocking signal applied to the charge transfer device; a pre-emphasis circuit in advance of the charge transfer device emphasizes the input signal in its higher-frequency region, and a de-emphasis circuit after the charge transfer device emphasizes the signal at the output of the device in the lower-frequency region of the signal to compensate for the emphasis provided by the pre-emphasis circuit so that the delay circuit has a total response that is substantially flat. Such a delay circuit can be included in a stereophonic sound extending circuit to delay, by several milliseconds, one of left- and right-channel signals applied to a matrix circuit which provides extended signal components formed by combining each one of the left- and ...

Stereo separation control

Improved circuitry for providing stereo separation control in audio playback systems wherein undesirable volume level change and undesirable signal degradation due to noise, distortion and crosstalk are substantially reduced or eliminated by utilizing, in a preferred embodiment, a precision difference amplifier, a signal inverter, a tapped separation control, and a crosstalk-cancelling resistor.

Audio reproducing system

High and low frequency components C_n1, C_n2 are generated by causing a center channel audio signal Cn to pass through a high-pass filter HPF and a low-pass filter LPF, and the high frequency component C_n1 is supplied to the center channel loudspeaker SP_C. Also, two low frequency components C_n2L, C_n2R, which have a different phase mutually, are generated by causing the low frequency component C_n2 to pass through phase shifters 2, 3 . An synthesized audio signal SL is generated by synthesizing the low frequency component C_n2L and the front-left channel audio signal FL, and then supplied to the front-left channel loudspeaker SP_L. An synthesized audio signal SR is generated by synthesizing the low frequency component C_n2R and the front-right channel audio signal FR, and is supplied to the front-right channel loudspeaker SP_R.

METHOD AND APPARATUS TO CREATE A SOUND FIELD

The invention generally relates to a method and apparatus for taking an input signal, replicating it a number of times and modifying each of the replicas before routing them to respective output transducers such that a desired sound field is created. This sound field may comprise a directed beam, focused beam or a simulated origin. In a first aspect, delays are added to sound channels to remove the effects of different travelling distances. In a second aspect, a delay is added to a video signal to account for the delays added to the sound channels. In a third aspect, different window functions are applied to each channel to give improved flexibility of use. In a fourth aspect, a smaller extent of transducers is used top output high frequencies than are used to output low frequencies. An array having a larger density of transducers near the center is also provided. In a fifth aspect, a line of elongate transducers is provided to give good directivity in a plane. In a sixth aspect, sound ...

Sound Localization for an Electronic Call

During an electronic call between two individuals, a sound localization point simulates a location in empty space from where an origin of a voice of one individual occurs for the other individual.

System and method for localization of sounds in three-dimensional space

A system and method for localization of sounds in 3-D space using Head Related Transfer Functions (HRTFs) are disclosed herein. A plurality of audio signals and/or sound sources, or "voices", represented by data, such as one or more audio files or streaming audio data, is transmitted from an audio source, such as a compact disc player, to an audio processing system. The audio processing system, in one embodiment, prioritizes the voices based on characteristics of the voices, such as the relative listening benefit of sound localization of a particular voice. In one embodiment, a finite impulse response (FIR) filter is used to apply a HRTF to the voices. The audio processing system distributes up to a fixed number of coefficients among the voices, where the distribution of the coefficients, in one embodiment, is based on the priority of each channel. Those voices having a higher priority than other voices could have a greater number of coefficients assigned to them than the lower priority ...

Stereo widening network for two loudspeakers

The invention relates to a method, a system, a module, an electronic device and to a computer program product for widening a two-channel input. Two audio channels are input and filtered by equalizing said channels. The filtered channels are mixed with their opposite channels in a cross-talk network and output from loudspeakers and by this providing a spatial impression for audio.

VIDEO/AUDIO SYSTEM AND METHOD ENABLING A USER TO SELECT DIFFERENT VIEWS AND SOUNDS ASSOCIATED WITH AN EVENT

A video/audio system includes an interface device that receives a plurality of audio and video signals from a plurality of sources. The interface device combines these signals into various combinations and transmits the combinations to a receiver. The receiver is configured to interface one of the combinations of signals with a user. In this regard, the receiver allows the user to select one of the combinations, and in response, the receiver separates the video signal(s) of the selected combination from the audio signal(s) of the selected combination. Then, the receiver renders the video signal(s) via a display device and produces a sound defined by the audio signal(s) via a speaker. Accordingly, the user is able to control which set of audio and video signals are interfaced with the user.

Apparatus and method for enhancing a spatial perception of an audio signal

An apparatus and a method for enhancing a spatial perception of an audio signal are provided creating increased interaural-level differences. To obtain this effect, two dipoles are used: one for producing a left audio signal and one for producing a right audio signal.

Audio System

An audio system comprises an audio source terminal 11 and a audio playback terminal 13, connected to each another by a wireless data link 14. The source terminal 11 comprises a source computer 15, and a cellular modem 17. The playback terminal 13 comprises a playback computer 19 having an internal processor 23 and an audio processor 24. Connected to the processor 23 is a cellular modem 21, an audio transducer 25, and a user control 27. Data relating to audio components, representing different services, is stored at the source terminal 11 where it is spatially processed and transmitted to the playback terminal. At the same time, each individual audio component is transmitted at a lower bit-rate than the spatially processed data, to the audio source terminal 11, whereafter it is spatially processed. Although the low bit-rate transmission causes a loss of audio quality, the positional data remains unaffected. Accordingly, when played, the combination of a high quality signal with low three-dimensional ...

Multi-channel Output Device for a Multimedia Device

A multi-channel output device for a multimedia device includes a plurality of reception ends, an amplifier having an input end and an output end, a plurality of sound output devices for outputting sound, a first switching unit coupled between the plurality of reception ends and the input end of the amplifier for coupling a reception end of the plurality of reception ends to the input end of the amplifier according to a control signal, a second switching unit coupled between the output end of the amplifier and the plurality of sound output devices for coupling the output end of the amplifier to a sound output device of the plurality of sound output devices according to the control signal, and a control unit coupled to the first switching unit and the second switching unit for outputting the control signal.

Display apparatus and controlling method thereof

A display apparatus includes a speaker, a storage configured to store first rendering type information and second rendering type information based on a volume level of each content genre; a display, and a processor configured to render a sound signal of a content based on at least one of the first rendering type information and the second rendering type information corresponding to a genre of the content on the display and output the rendered sound signal through the speaker, wherein the processor, based on a volume level which is currently set in the display apparatus being within a predetermined range, renders the sound signal based on third rendering type information which is obtained by applying a weight according to the set volume level to at least one of the first rendering type information and the second rendering type information.

Signal processing device and signal processing method

The present disclosure relates to a signal processing device, a signal processing method, and a program that make it possible to readily achieve personalization of head-related transfer functions in all bands. A synthesis unit generates a third head-related transfer function by synthesizing a characteristic of a first band extracted from a first head-related transfer function of a user and a characteristic of a second band other than the first band extracted from a second head-related transfer function measured in a second measurement environment different from a first measurement environment in which the first head-related transfer function is measured. The present disclosure may be applied to, for example, a mobile terminal such as a smartphone.

Virtual source localization of audio signal

An object is to reduce the signal processing volume in an audio processing unit. In an audio signal processing method which performs virtual acoustic image localization processing for sound source signals having at least one type of information among position information, movement information, and localization information, based on this information, when there are a plurality of changes in this information within a prescribed time unit, a single information change is generated based on this plurality of information changes, and virtual acoustic image localization processing is performed for the sound source signals based on this generated information change.

Method and apparatus for decomposing a stereo recording using frequency-domain processing employing a spectral subtractor

An apparatus for generating a stereo side signal having a first side channel and a second side channel from a stereo input signal having a first input channel and a second input channel is provided. The apparatus comprises a modification information generator (110) for generating modification information based on mid-side information. Furthermore, the apparatus comprises a signal manipulator (120) being adapted to manipulate the first input channel based on the modification information to obtain the first side channel and being adapted to manipulate the second input channel based on the modification information to obtain the second side channel. The modification information generator (110) comprises a spectral subtractor (115) for generating the modification information by generating a difference value indicating a difference between a mono mid signal or a mono side signal of the stereo input signal and the first or the second input channel.

AN AUDIO STEREO PROCESSING METHOD, DEVICE AND SYSTEM

Reservation procedure in digital transmission networks and digital switch

The method involves reserving transmission capacities and selecting requests for data streams with different bandwidth to be transferred in digital data transmission networks with a maximum transmission rate, whereby a certain transmission capacity is available, which is smaller or equal to the maximum transmission rate, certain bandwidths or bandwidth groups use determined data transmission rates, and certain data transmission rates are reserved for certain bandwidths or bandwidth groups. A request for the transmission of data with a certain bandwidth or bandwidth group is only then accepted, if predetermined criteria are fulfilled with respect to the usage of the data network, only if the unused data transmission capacity after the acceptance of the transmission is larger or equal to the sum of the reserved data transmission rates with the exception of the reserved data transmission rate for the requesting bandwidth or bandwidth group.

APPARATUS FOR PROCESSING STEREOPHONIC SIGNALS

In order to simplify filter construction in a circuit which can be used for crosstalk cancellation in binaural signals to be played through loudspeakers, the circuit includes first and second signal paths for receiving as inputs, left and right binaural signals LEFT-IN, RIGHT-IN. The first signal path (14) includes a first summing junction (10) and the second signal path includes a second summing junction (12). The output of summing junction (10) is coupled by a first cross-path (14) to an input of summing junction (12), and the output of summing junction (12) is coupled by a second cross-path (22) to an input of junction (10). The first and second cross-paths include respective first and second filter means (17, 25) which include crosstalk filters (18, 26) each having a transfer function A/S, where A and S represent far-ear and near-ear HRTFs. The outputs of the summing junctions (10, 12) represent output signals LEFT-OUT, RIGHT-OUT, incorporating crosstalk cancellation.

Control circuit for a spatial stereo expander

Sound field and sound image control apparatus and method

APPARATUS, METHOD OR COMPUTER PROGRAM FOR GENERATING AN OUTPUT DOWNMIX REPRESENTATION

両耳間時間遅延復元システム及び方法

СИСТЕМА И СПОСОБ ДЛЯ ОБРАБОТКИ ЗВУКА

Изобретение относится к средствам обработки звука. Технический результат заключается в улучшении пространственного восприятия звукового сигнала. Система обработки звука принимает стереосигнал, который с помощью блока сегментации делится на частотно-временные сегменты стереосигнала, каждый из которых может соответствовать выборке частотной области в данном временном сегменте. Блок разложения разлагает частотно-временные сегменты сигнала, для каждой пары частотно-временных сегментов стереосигнала, путем осуществления этапов: определения меры подобия, указывающей степень подобия частотно-временных сегментов стереосигнала; генерации частотно-временного сегмента суммарного сигнала, как суммы частотно-временных сегментов стереосигнала; и генерации центрального частотно-временного сегмента сигнала из частотно-временного сегмента суммарного сигнала и пары боковых частотно-временных сегментов стереосигнала из пары частотно-временных сегментов стереосигнала в соответствии с мерой подобия. Затем генератор ...

ГЕНЕРАЦИЯ ДЕКОРРЕЛИРОВАННЫХ СИГНАЛОВ

Изобретение относится к устройству и способу генерации декоррелированных сигналов и, более конкретно, к возможности получения декоррелированных сигналов из сигнала, содержащего переходные процессы. В случае переходных аудиовходных сигналов, при многоканальном аудиовоспроизведении, некоррелированные выходные сигналы генерируются из аудиовходного сигнала таким образом, что аудиовходной сигнал смешивается с представлением аудиовходного сигнала, задержанного на время задержки так, что в первом временном интервале первый выходной сигнал соответствует аудиовходному сигналу, и второй выходной сигнал соответствует задержанному представлению аудиовходного сигнала, причем во втором временном интервале первый выходной сигнал соответствует задержанному представлению аудиовходного сигнала, и второй выходной сигнал соответствует аудиовходному сигналу. Технический результат - улучшение качества сигнала в присутствии переходных сигналов. 4 н. и 22 з.п. ф-лы, 10 ил.

СПОСОБ ПОЛУЧЕНИЯ И ЛОКАЛИЗАЦИИ КАЖУЩЕГОСЯ ИСТОЧНИКА ЗВУКА В ТРЕХМЕРНОМ ПРОСТРАНСТВЕ И СИСТЕМА ДЛЯ ЕГО ОСУЩЕСТВЛЕНИЯ

Использование: для получения и локализации кажущегося источника звука в трехмерном пространстве. Сущность изобретения: разделяют электрический сигнал на соответствующие сигналы первого и второго каналов, производят одновременное независимое изменение амплитуд и сдвиг фаз, подаваемых на первый и второй громкоговорители сигналов в зависимости от их частоты, поддерживают разность фаз и амплитуд сигналов первого и второго каналов в последовательных дискретных частотных полосах звукового диапазона. При этом каждый последующий сдвиг фазы отличен от предыдущего. 2 с. и 5 з.п. ф-лы, 24 ил.

УСТРОЙСТВО ОБРАБОТКИ АУДИОСИГНАЛА И СПОСОБ СНИЖЕНИЯ ПЕРЕКРЕСТНЫХ ПОМЕХ АУДИОСИГНАЛА

Изобретение относится к средствам обработки аудиосигнала. Технический результат заключается в повышении эффективности фильтрации входного аудиосигнала. Разбивают входной аудиосигнал (L) левого канала на первый входной аудиоподсигнал левого канала и второй входной аудиоподсигнал левого канала. Разбивают входной аудиосигнал (R) правого канала на первый входной аудиоподсигнал правого канала и второй входной аудиоподсигнал правого канала. Причем первый входной аудиоподсигнал левого канала и первый входной аудиоподсигнал правого канала распределены в первый предопределенный частотный диапазон. Второй входной аудиоподсигнал левого канала и второй входной аудиоподсигнал правого канала распределены во второй предопределенный частотный диапазон. Снижают перекрестные помехи между первым входным аудиоподсигналом левого канала и первым входным аудиоподсигналом правого канала в первом предопределенном частотном диапазоне для получения первого выходного аудиоподсигнала левого канала и первого выходного ...

АППАРАТУРА И МЕТОД МНОГОКАНАЛЬНОГО ПАРАМЕТРИЧЕСКОГО ПРЕОБРАЗОВАНИЯ

... 1. Многоканальный преобразователь параметра для генерирования параметра выходного уровня, который указывает на энергетическое соотношение между первым звуковым сигналом и вторым звуковым сигналом при воспроизведении многоканального пространственного звукового сигнала; характеризующийся тем, что он содержит преобразователь параметра объекта для обеспечения объектных параметров для множества звуковых объектов, подключенный к каналу понижающего микширования в зависимости от объектных звуковых сигналов, связанных со звуковыми объектами; при этом параметры объекта содержат энергетический параметр для каждого звукового объекта, определяющий энергетическую информацию сигнала звукового объекта; и генератор параметра для получения параметра выходного уровня с помощью комбинирования энергических параметров и параметров воспроизведения объекта, которые связаны с конфигурацией воспроизведения. ! 2. Многоканальный преобразователь параметра по п.1, характеризующийся тем, что он выполнен с возможностью ...

СПОСОБ И УСТРОЙСТВО ДЛЯ ВОСПРОИЗВЕДЕНИЯ ТРЕХМЕРНОГО ЗВУКОВОГО СОПРОВОЖДЕНИЯ

... 1. Способ воспроизведения стереофонического звукового сопровождения, при этом, способ содержит следующие этапы:получают информацию о глубине изображения, указывающую расстояние между, по меньшей мере, одним объектом в сигнале изображения и опорным местоположением;получают информацию о глубине звука, указывающую расстояние между, по меньшей мере, одним звуковым объектом в звуковом сигнале и опорным местоположением, на основании информации о глубине изображения; иобеспечивают звуковую перспективу для, по меньшей мере, одного звукового объекта на основании информации о глубине звука.2. Способ по п. 1, в котором этап получения информации о глубине звука содержит следующие этапы:получают максимальное значение глубины для каждой секции изображения, которая составляет сигнал изображения; иполучают значение глубины звука для, по меньшей мере, одного звукового объекта на основании максимального значения глубины.3. Способ по п. 2, в котором этап получения значения глубины звука содержит определение ...

Elektro-akustischer Wandler

DIE EINHUELLUNGSMODULATION BESEITIGENDES TONUEBERTRAGUNGSGERAET

STEREOWIEDERGABEANLAGE

Spatial audio capture, transmission and reproduction

An apparatus and method comprising, receiving at least two audio signals 201, before determining at least one lower frequency effect (LFE) parameter and at least one transport audio signal 203 based on the at least audio signals. This is followed by controlling the transmission/storage of the transport audio signal and the LFE parameter such that a rendering based on the transport signal and the LFE parameter enables a determination of at least one LFE channel. Preferably at least one spatial metadata parameter is determined. This spatial meta-data comprises at least one of a direction parameter associated with one frequency band, or a direct-to-total energy ratio associated with at least one frequency band. The means for determining the transport audio signal preferably comprise at least one of a down mix of the audio signals, a selection of the audio signals, an audio processing of the audio signals or an ambisonic audio processing. The at least two audio signals are at least one of a ...

A method of improving 3D sound reproduction

Mobile electronic device and sound playback method thereof

A mobile electronic device and a sound playback method thereof are provided. The mobile electronic device includes a sensor, a speaker, and a controller coupled to the sensor and the speaker. The sensor detects whether the speaker is blocked or not. When the speaker is blocked, the controller multiplies a sound signal by a transfer function and then outputs the multiplied sound signal. The speaker plays the sound signal outputted by the controller. The transfer function changes the direction in which the speaker plays the sound signal.

Audio system, audio signal processing device and method, and program

An audio system including a first speaker and a second speaker that are arranged in front of a predetermined listening position to be substantially bilaterally symmetrical with respect to the listening position; a third speaker and a fourth speaker that are arranged in front of the predetermined listening position to be substantially bilaterally symmetrical with respect to the listening position; a first attenuator that attenuates components that are equal to or less than a predetermined first frequency of an input audio signal; and an output controller that outputs sounds that are based on the input audio signal from the first speaker and the second speaker and outputs sounds that are based on the first audio signal in which components that are equal to or less than the first frequency of the input audio signal from the third speaker and the fourth speaker.

Immersive audio rendering system

A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

Directional Electroacoustical Transducing

A multichannel audio system for radiating sound to a listening area that includes a plurality of listening spaces. The audio system includes directional audio devices, positioned in a first of the listening spaces, close to a head of a listener, for radiating first sound waves corresponding to components of one of the channels and nondirectional audio devices, positioned inside the listening area and outside the listening space, distant from the listening space, for radiating sound waves corresponding to components of a second of the channels.

Sound processing apparatus and parameter setting method

A sound processing apparatus includes a processing unit that is configured to acquire an audio signal, perform a correction processing on the acquired audio signal and output the correction-processed audio signal to a sound emitting unit. The correction processing includes an indirect sound adjusting processing in which a given signal processing is performed on an audio signal so as to adjust an influence of an indirect sound to be heard at a sound receiving point, and a frequency characteristic adjusting processing in which a frequency characteristic of an audio signal is adjusted. In the correction processing, a frequency characteristic for the frequency characteristic adjusting processing is determined based on a frequency characteristic of the indirect sound adjusting processing.

Method and apparatus for reproducing three-dimensional sound

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

UPMIXING METHOD AND SYSTEM FOR MULTICHANNEL AUDIO REPRODUCTION

An audio signal enhancing device, and a corresponding method of enhancing stereophonic signals, is provided which generates an enhanced signal with improved spatial sound image quality for upmixing a stereophonic input signal. When used in combination with a centre channel processor or LFE processor, an improved processing of the input signals is provided resulting in final centre channel and at least one LFE sub-woofer channel wherein the problems and disadvantages of the prior art are resolved. The result is a centre and LFE signal that contains a stable, non time-smeared image with a high quality natural-sounding fidelity. These advantages are achieved especially for time-delayed or phase-panned stereo input signals, independently of whether they are matrix encoded or non-matrix encoded input signals. 1. An audio signal enhancing device for upmixing a stereophonic input signal comprising two audio signals , the device comprising:signal enhancement means for processing the two input signals to generate at least one enhanced signal, the signal enhancement means comprising:two parallel processing lines comprising two parallel processing branches each;the first processing branch comprising adaptive filter means; andthe second processing branch comprising means for delaying a signal;means for combining the output signal of the first processing branch of the first processing line with the output signal of the second processing branch of the second processing line to generate a first enhanced signal;means for combining the output signal of the first processing branch of the second processing line with the output signal of the second processing branch of the first processing line to generate a second enhanced signal; andmeans for combining the first and second enhanced signals to generate a third enhanced signal; andcontrol means for controlling the signal enhancement means.2. The audio signal enhancing device of claim 1 , wherein the control means is adapted:to analyse ...

METHODS AND APPARATUS TO GENERATE SPATIAL AUDIO BASED ON COMPUTER VISION

Methods, apparatus, systems, and articles of manufacture are disclosed to generate spatial audio based on computer vision. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to determine a position of an audio source based on an image generated via a camera, and apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source. 1. An apparatus comprising:at least one memory;instructions in the apparatus; and determine a position of an audio source based on an image generated via a camera; and', 'apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source., 'processor circuitry to execute the instructions to2. The apparatus of claim 1 , wherein the processor circuitry is to determine the position of the audio source as an azimuth angle.3. The apparatus of claim 1 , wherein the audio source is a person speaking.4. The apparatus of claim 3 , wherein the processor circuitry is to:detect a face of the person speaking in the image; andgenerate a bounding box of the face of the person speaking in the image.5. The apparatus of claim 4 , wherein the processor circuitry is to:determine a lateral location within the image of the person speaking based on a location of the bounding box; anddetermine a proximity of the person speaking with respect to the camera based on a size of the bounding box.6. The apparatus of claim 5 , wherein the processor circuitry is to determine the position of the person speaking based on the lateral location and the proximity.7. The apparatus of claim 5 , wherein the processor circuitry is to determine the proximity of the person speaking to be on a near plane claim 5 , a medium plane claim 5 , or a far plane.8. The apparatus of claim 1 , wherein the microphone is located at the audio source.9. The apparatus of claim 1 , wherein the audio ...

AUDIO PROCESSING

According to an example embodiment, a technique for processing an input audio signal () comprising a multi-channel audio signal is provided, the technique comprising: deriving (), based on the input audio signal (), a first signal component (-) comprising a multi-channel audio signal that represents a focus portion of a spatial audio image conveyed by the input audio signal and a second signal component (-) comprising a multi-channel audio signal that represents a non-focus portion of the spatial audio image; processing () the second signal component (-) into a modified second signal component () wherein the width of the spatial audio image is extended from that of the second signal component (-); and combining () the first signal component (-) and the modified second signal component () into an output audio signal () comprising a multi-channel audio signal that represents partially extended spatial audio image. 1. An apparatus for processing an input audio signal comprising a multi-channel audio signal , the apparatus comprising at least one processor; and at least one memory including computer program code , which when executed by the at least one processor , causes the apparatus to:derive, based on the input audio signal, a first signal component comprising a multi-channel audio signal that represents a focus portion of a spatial audio image conveyed by the input audio signal and a second signal component comprising a multi-channel audio signal that represents a non-focus portion of the spatial audio image;process the second signal component into a modified second signal component wherein the width of the spatial audio image is extended from that of the second signal component; andcombine the first signal component and the modified second signal component into an output audio signal comprising a multi-channel audio signal that represents a partially extended spatial audio image.2. An apparatus according to claim 1 , wherein the apparatus is caused to derive the ...

Converting Multi-Microphone Captured Signals To Shifted Signals Useful For Binaural Signal Processing And Use Thereof

A method includes, estimating directional information based on multiple input channel signals representing at least one arriving sound from a sound source captured by respective multiple microphones that have respective known locations relative to each other, wherein said estimating comprises finding a time delay that removes a time difference between said first and second input channel signals; deriving a mid-signal and a side signal on a basis of a first input channel signal, a second input channel signal and said estimated directional information; and generating an output signal comprising a plurality of output channels using said mid-signal, said side signal and said estimated directional information such that the output signal retains a spatial representation of the captured at least one arriving sound. Apparatus and program products are also disclosed. 1. A method comprising:estimating directional information based on multiple input channel signals representing at least one arriving sound from a sound source captured by respective multiple microphones that have respective known locations relative to each other, wherein said estimating comprises finding a time delay that removes a time difference between said first and second input channel signals;deriving a mid-signal and a side signal on a basis of a first input channel signal, a second input channel signal and said estimated directional information; andgenerating an output signal comprising a plurality of output channels using said mid-signal, said side signal and said estimated directional information such that the output signal retains a spatial representation of the captured at least one arriving sound.2. The method as claimed in claim 1 , wherein said deriving comprises; deriving the mid-signal as a sum of one of said first and second input channel signals shifted by said time delay and the other one of said first and second input channel signals; and deriving the side signal as a difference between the ...

Modified Directional Effect

An example method is performed by a media playback system comprising a plurality of audio drivers configured to output audio content according to a first radiation pattern that produces an inherent directional effect. Based on data representing positions of one or more listeners in a listening area, the system determines first and second transfer functions corresponding to the first and second audio drivers, respectively. One or both of the transfer functions configure the first and second audio drivers to output audio content according to a second radiation pattern that produces a modified directional effect relative to the first radiation pattern. The system applies the transfer function to audio content thereby causing the first and second audio drivers to play back audio content according to the second radiation pattern. 1. A media playback system comprising:a processor;a network interface;a first audio driver and a second audio driver, wherein the first and second audio drivers are configured to output audio content according to a first radiation pattern that produces an inherent directional effect; and receiving, via the network interface, first data representing a first channel of multi-channel audio content and second data representing a second channel of the multi-channel audio content, wherein the first and second data comprises audio data over a range of frequencies;', 'receiving third data comprising positions of one or more listeners in a listening area in which the media playback system operates;', 'determining, based on the third data, a first transfer function corresponding to the first audio driver and a second transfer function corresponding to the second audio driver, wherein at least one of the first and second transfer functions configure the first and second audio drivers to output audio content according to a second radiation pattern that produces a modified directional effect relative to the first radiation pattern;', 'generating a first ...

METHOD FOR PROCESSING SOUND ON BASIS OF IMAGE INFORMATION, AND CORRESPONDING DEVICE

A method of processing an audio signal including at least one audio object based on image information includes: obtaining the audio signal and a current image that corresponds to the audio signal; dividing the current image into at least one block; obtaining motion information of the at least one block; generating index information including information for giving a three-dimensional (3D) effect in at least one direction to the at least one audio object, based on the motion information of the at least one block; and processing the audio object, in order to give the 3D effect in the at least one direction to the audio object, based on the index information. 1. A method of processing an audio signal comprising at least one audio object based on image information , the method comprising:obtaining the audio signal and a current image that corresponds to the audio signal;dividing the current image into at least one block;obtaining motion information of the at least one block;generating index information comprising information for giving a three-dimensional (3D) effect in at least one direction to the at least one audio object, based on the motion information of the at least one block;processing the audio object, in order to give the 3D effect in the at least one direction to the audio object, based on the index information; andoutputting the audio signal including the processed the audio object via a speaker.2. The method of claim 1 , wherein the generating of the index information comprisesobtaining motion information of the current image based on the motion information about the at least one block, and generating the index information based on the motion information of the current image.3. The method of claim 1 , wherein the obtaining of the motion information of the at least one block comprises:determining a block, having a lowest pixel value difference from each block of the current image, from among at least one block that is included in an image that is prior or ...

APPARATUS AND METHOD FOR PROVIDING A MEASURE OF SPATIALITY ASSOCIATED WITH AN AUDIO STREAM

Apparatus for evaluating an audio stream, wherein the audio stream includes audio channels to be reproduced at at least two different spatial layers, wherein the two spatial layers are arranged in a manner distanced along a spatial axis. The apparatus is configured to evaluate the audio channels of the audio stream so as to provide a measure of spatiality associated with the audio stream. 1. An apparatus for evaluating an audio stream ,wherein the audio stream comprises audio channels to be reproduced at at least two different spatial layers, wherein the two spatial layers are arranged in a manner distanced along a spatial axis,wherein the apparatus is configured to evaluate the audio channels of the audio stream as to provide a measure of spatiality associated with the audio stream.2. An apparatus according to claim 1 , wherein the spatial axis is oriented horizontally claim 1 , or wherein the spatial axis is oriented vertically.3. An apparatus according to claim 1 , wherein the apparatus is configured to acquire a first level information based on a first set of audio channels of the audio stream and to acquire a second level information based on a second set of audio channels of the audio stream claim 1 , andwherein the apparatus is configured to determine a spatial level information based on the first level information and the second level information and to determine the measure of spatiality based on the spatial level information.4. An apparatus according to claim 3 , wherein the first set of audio channels of the audio stream is disjoint to the second set of audio channels of the audio stream.5. An apparatus according to claim 3 , wherein the first set of audio channels of the audio stream is to be reproduced on loudspeakers in one or more first spatial layers and wherein the second set of audio channels of the audio stream is to be reproduced on loudspeakers on one or more second spatial layers claim 3 ,wherein the one or more first layers and the one or more ...

Method that Expedites Playing Sound of a Talking Emoji

A method expedites playing sound of a talking emoji from a first person with a first portable electronic device (PED) to a second person with a second PED. The second PED receives the talking emoji in mono sound and convolves the mono sound into binaural sound before receiving a request to play the sound to the second user. The second PED then plays the sound of the talking emoji in binaural sound after receiving the request from the second user.

SOUND SIGNAL PROCESSING DEVICE AND SOUND SIGNAL PROCESSING METHOD

A sound signal processing device includes: a vocal remover which generates a first output signal based on first-channel and second-channel sound signals and a first coefficient indicating a vocal bandwidth to be removed; a surround sound processor which generates a second output signal by adding a surround sound effect to the first output signal; an amplifier which amplifies a signal at an amplification factor that is based on a second coefficient; a synthesizer which synthesizes the second output signal with one of the first-channel and second-channel sound signals, and synthesizes a signal that is the second output signal inverted with another one of the first-channel and second-channel sound signals; and a coefficient determination unit which sets the second coefficient such that the amplification factor, used when the vocal bandwidth to be removed is greater than a first bandwidth, is greater than the amplification factor for the first bandwidth. 1. A sound signal processing device , comprising:a remover which generates a first output signal based on a first-channel sound signal, a second-channel sound signal, and a first coefficient indicating a vocal bandwidth to be removed, the first output signal being a signal from which a vocal component has been removed;a surround sound processor which generates a second output signal by adding a surround sound effect to the first output signal;an amplifier which amplifies an input signal at an amplification factor that is based on a second coefficient, the amplifier being (i) connected upstream of the remover, (ii) connected between the remover and the surround sound processor, (iii) included in the remover, or (iv) included in the surround sound processor;a first synthesizer which synthesizes the second output signal with one of the first-channel sound signal and the second-channel sound signal;a second synthesizer which synthesizes a signal that is the second output signal inverted with another one of the first-channel ...

LOUDNESS ADJUSTMENT FOR DOWNMIXED AUDIO CONTENT

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content. 1. A method for gain adjusting audio signals based on encoder-generated loudness metadata , the method comprising:receiving, by an audio decoder operating in a playback channel configuration different from a reference channel configuration, an audio signal for the reference channel configuration, the audio signal including audio sample data for each channel of the reference channel configuration, and the encoder-generated loudness metadata, the encoder-generated loudness metadata comprising loudness metadata for a plurality of channel configurations including the playback channel configuration and the reference channel configuration;selecting from the loudness metadata for the plurality of channel configurations, the loudness metadata for the playback channel configuration;determining loudness adjustment gains from the loudness metadata for the playback channel configuration; andapplying the loudness adjustment gains as a part of overall gains applied to the audio sample data to generate output audio sample data for each channel of the playback channel configuration;wherein the loudness adjustment gains depend on a loudness level indicated by the loudness metadata for the playback channel configuration and a reference loudness level.2. The method of claim 1 , wherein the overall gains comprise one or more of: ...

Method and Device for Processing Audio Signal

The embodiments of the present disclosure disclose a method and device for processing an audio signal. A specific embodiment of the method includes acquiring a head image of a target user and an audio signal to be processed; determining head attitude angles of the target user based on the head image, and determining a distance between a target sound source and the head of the target user; and inputting the head attitude angles, the distance and the audio signal to be processed into a preset head related transfer function to obtain a processed left channel audio signal and a processed right channel audio signal, wherein the head related transfer function is used to characterize a correspondence between the head attitude angles, the distance and the audio signal to be processed, and the processed left channel audio signal and the processed right channel audio signal. 1. A method for processing an audio signal , comprising:acquiring a head image of a target user and an audio signal to be processed;determining head attitude angles of the target user based on the head image, and determining a distance between a target sound source and the head of the target user; andinputting the head attitude angles, the distance and the audio signal to be processed into a preset head related transfer function to obtain a processed left channel audio signal and a processed right channel audio signal, wherein the head related transfer function is used to characterize a correspondence between the head attitude angles, the distance and the audio signal to be processed, and the processed left channel audio signal and the processed right channel audio signal.2. The method according to claim 1 , wherein determining the head attitude angles of the target user based on the head image comprises:inputting the head image into a pre-trained head attitude recognition model to obtain the head attitude angles of the target user, wherein the head attitude recognition model is used to characterize a ...

Binaural Dialogue Enhancement

Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation. 1. A method for dialogue enhancing audio content having one or more audio components , wherein each component is associated with a spatial location , comprising:providing a first audio signal presentation of the audio components intended for reproduction on a first audio reproduction system;providing a second audio signal presentation of the audio components intended for reproduction on a second audio reproduction system;receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation;applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; andcombining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system;wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.2. The method according to claim 1 , wherein both said first and second audio signal presentations are binaural audio signal presentations.3. The method according to claim 1 , wherein only ...

METHOD AND SYSTEM FOR HEAD-RELATED TRANSFER FUNCTION GENERATION BY LINEAR MIXING OF HEAD-RELATED TRANSFER FUNCTIONS

A method for performing linear mixing on coupled Head-related transfer functions (HRTFs) to determine an interpolated HRTF for any specified arrival direction in a range (e.g., a range spanning at least 60 degrees in a plane, or a full range of 360 degrees in a plane), where the coupled HRTFs have been predetermined to have properties such that linear mixing can be performed thereon (to generate interpolated HRTFs) without introducing significant comb filtering distortion. In some embodiments, the method includes steps of: in response to a signal indicative of a specified arrival direction, performing linear mixing on data indicative of coupled HRTFs of a coupled HRTF set to determine an HRTF for the specified arrival direction; and performing HRTF filtering on an audio input signal using the HRTF for the specified arrival direction.

IMMERSIVE AUDIO RENDERING SYSTEM

A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect. 127-. (canceled)28. A method of processing audio signals , the method comprising:receiving left and right audio signals, the left and right audio signals each comprising information about a spatial position of a sound source relative to a listener;calculating depth steering information using the left and right audio signals;decorrelating the left and right audio signals by an amount that depends at least partly on the depth steering information to produce decorrelated left and right audio signals;calculating difference information in the decorrelated left and right audio signals;applying at least one perspective filter to the difference information to produce left and right output signals; andproviding the left and right output signals for playback,wherein the method is performed by one or more hardware processors.29. The method of claim 28 , further comprising performing at least one of:detecting an envelope of the difference information or smoothing the difference information.30. The method of claim 29 , further comprising modulating the application of the at least one perspective filter based at least in part on one or both of the envelope of the difference information and the smoothed difference information.31. The method of claim 28 , further comprising normalizing the difference information based at least in part on signal levels of ...

AUDIO PROCESSING

A method comprising: dividing a virtual space using virtual partitions that affect perception of the virtual space by a user within the virtual space; in response to a first action in the virtual space relative to a first virtual partition by a user making a first change to how the first virtual partition affects the virtual space perceived by the user. 127-. (canceled)28. An apparatus comprising:at least one processor; andat least one memory comprising computer program code,the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus to perform at least the following:divide a virtual space using one or more virtual partitions that affect rendering of the virtual space to affect perception of the virtual space by a user within the virtual space;in response to a first action in the virtual space relative to at least a first virtual partition by a user make a first change to how at least the first virtual partition affects the virtual space perceived by the user.29. The apparatus of claim 28 , wherein the one or more virtual partitions define a plurality of scenes demarcated at least partly by the one or more virtual partitions claim 28 , wherein a scene is at least partially separated from an adjacent scene by at least one partition and wherein each scene comprises different sound objects.30. The apparatus of further caused to render audio associated with a current scene to the user at a greater volume than audio associated with any other scene claim 29 , wherein the current scene is determined by a position of a user in the virtual space claim 29 , the current scene being the scene within which the user is currently located.31. The apparatus of further caused to render audio associated with a first scene different to the current scene to the user at an increased volume in dependence upon an increased user proximity to a virtual partition dividing the current scene from the first scene.32. The apparatus of ...

Binaural Dialogue Enhancement

Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation. 1. A method of processing immersive audio content , comprising:receiving a first audio signal presentation of the immersive audio content, the first audio signal presentation configured to reproduce on a first audio reproduction system;receiving a second audio signal presentation of the immersive audio content, the second audio signal presentation configured to reproduce on a second audio reproduction system;receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation;forming a dialogue presentation of the dialogue components by applying the set of dialogue estimation parameters to the first audio signal presentation; andcombining the dialogue presentation with the second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system,wherein at least one of the first or second audio signal presentation is a binaural audio signal presentation.2. The method of claim 1 , wherein the immersive audio content includes one or more spatial audio components.3. The method of claim 1 , wherein both said first and second audio signal presentations are ...

Blind Detection of Binauralized Stereo Content

An apparatus and method of blind detection of binauralized audio. If the input content is detected as binaural, a second binauralization may be avoided. In this manner, the user experience avoids audio artifacts introduced by multiple binauralizations. 1. An apparatus for detecting binauralization of an audio signal , the apparatus comprising:a feature extractor that is configured to receive an input audio signal, and that is configured to extract a plurality of features from the input audio signal;a classifier that is configured to receive the plurality of features extracted by the feature extractor, that is configured to perform a classification of the plurality of features, and that is configured to generate a result based on the classification, wherein the result indicates a probability that the input audio signal is a binaural signal versus a stereo signal;a virtualizer that is configured to receive the input audio signal, and that is configured to perform virtualization on the input audio signal to generate a binaural audio signal; anda mixer that is configured to receive the input audio signal, the binaural audio signal and the result, and that is configured to mix the input audio signal and the binaural audio signal, according to the result, to generate an output audio signal,wherein when the result indicates that the input audio signal is the stereo signal, the output audio signal corresponds to the binaural audio signal, andwherein when the result indicates that the input audio signal is the binaural signal, the output audio signal corresponds to the input audio signal.2. The apparatus of claim 1 , further comprising:a smoother that is configured to smooth the result from the classifier, and that is configured to provide the result, having been smoothed, to the mixer,wherein the mixer is configured to mix the input audio signal and the binaural audio signal according to the result having been smoothed by the smoother.3. The apparatus of claim 2 , wherein ...

ELECTRONIC APPARATUS, CONTROL METHOD THEREOF AND COMPUTER PROGRAM PRODUCT USING THE SAME

An electronic apparatus includes a memory configured to store instructions; and a processor configured to execute the instructions to: separate an input audio signal into a plurality of channel signals; identify a gain corresponding to a feature difference between a first channel signal, from among the plurality of channel signals, and a second channel signal, from among the plurality of channel signals; and adjust relative ratios among a plurality of output signals according to the identified gain to generate an output audio signal in which a sound image is varied. 1. An electronic apparatus comprising:a signal receiver configured to receive an audio signal;an output interface configured to be output an audio signal; and separate the received audio signal into a plurality of channel signals;', 'identify a gain corresponding to a feature difference between a first channel signal, from among the plurality of channel signals, and a second channel signal, from among the plurality of channel signals; and', 'control to adjust relative ratios among a plurality of output signals according to the identified gain to change a sound image of the audio signal outputted through the output interface., 'a processor configured to2. The electronic apparatus according to claim 1 , wherein the processor is further configured to adjust a relative ratio between the first channel signal and the second channel signal generated from the separated plurality of channel signals.3. The electronic apparatus according to claim 1 , wherein the feature difference comprises a phase difference between the first channel signal and the second channel signal.4. The electronic apparatus according to claim 1 , wherein the processor is further configured to:convert the first channel signal and the second channel signal into frequency domains; andidentify a feature difference between the first channel signal and the second channel signal, converted into the frequency domains.5. The electronic apparatus ...

METHOD OF PROVIDING AUDIO IN A VEHICLE, AND AN AUDIO APPARATUS FOR A VEHICLE

A method for providing audio in a vehicle includes: providing a plurality of audio channels comprising a first audio channel and a second audio channel; processing, by an audio signal processing system, each audio channel of said plurality of audio channels to establish a principal signal and an accommodating signal for each audio channel of said plurality of audio channels; supplying said principal signal of said first audio channel to a forward loudspeaker of a loudspeaker pair and supplying said accommodating signal of said first audio channel to a backward loudspeaker of the loudspeaker pair to generate a forward directional radiation pattern based on said first audio channel; supplying said principal signal of said second audio channel to said backward loudspeaker of said loudspeaker pair and supplying said accommodating signal of said second audio channel to said forward loudspeaker of said loudspeaker pair to generate a backward directional radiation pattern based on said second audio channel; and guiding one or both of said forward directional radiation pattern and said backward directional radiation pattern to one or more locations in said vehicle. 1. A method for providing audio in a vehicle , said method comprising the steps of:providing a plurality of audio channels comprising a first audio channel and a second audio channel;processing, by an audio signal processing system, each audio channel of said plurality of audio channels to establish a principal signal and an accommodating signal for each audio channel of said plurality of audio channels;supplying said principal signal of said first audio channel to a forward loudspeaker of a loudspeaker pair and supplying said accommodating signal of said first audio channel to a backward loudspeaker of the loudspeaker pair to generate a forward directional radiation pattern based on said first audio channel;supplying said principal signal of said second audio channel to said backward loudspeaker of said ...

METHOD FOR GENERATING BINAURAL SIGNALS FROM STEREO SIGNALS USING UPMIXING BINAURALIZATION, AND APPARATUS THEREFOR

Disclosed is an audio signal processing method including: receiving a stereo signal; transforming the stereo signal into a frequency-domain signal; rendering the first signal based on a first ipsilateral filter coefficient; generating a frontal ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a second ipsilateral filter coefficient; generating a side ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a contralateral filter coefficient; generating a side contralateral signal relating to the frequency-domain signal; transforming an ipsilateral signal, generated by mixing the frontal ipsilateral signal and the side ipsilateral signal, and the side contralateral signal into a time-domain ipsilateral signal and a time-domain contralateral signal, which are time-domain signals, respectively; and generating a binaural signal by mixing the time-domain ipsilateral signal and the time-domain contralateral signal. 116-. (canceled)17. An audio signal processing method comprising:receiving a virtual speaker layout,wherein the virtual speaker layout includes a plurality of virtual speakers,wherein the virtual speaker layout comprises of a plurality of cone of confusion (COC)s;obtaining ipsilateral signals on a frequency domain for each of signals of the plurality of virtual speakers;obtaining contralateral signals on the frequency domain for each of signals of the plurality of virtual speakers;obtaining a mixed ipsilateral signal by mixing the ipsilateral signals on the frequency domain;obtaining a plurality of mixed contralateral signals by mixing contralateral signals of virtual speakers located in a same CoC among the contralateral signals on the frequency domain;obtaining an ipsilateral signal on a time domain by converting the mixed ipsilateral signal;obtaining a plurality of contralateral signals on the time domain by converting the plurality of mixed contralateral signals; ...

Stereo separation and directional suppression with omni-directional microphones

Systems and methods for stereo separation and directional suppression are provided. An example method includes receiving a first audio signal, representing sound captured by a first microphone associated with a first location, and a second audio signal, representing sound captured by a second microphone associated with a second location. The microphones comprise omni-directional microphones. The distance between the first and second microphones is limited by the size of a mobile device. A first channel signal of a stereo signal is generated by forming, based on the first and second audio signals, a first beam at the first location. A second channel signal of the stereo signal is generated by forming, based on the first and second audio signals, a second beam at the second location. First and second directions, associated respectively with the first and second beams, are fixed relative to a line between the first and second locations.

Novel Method for Reducing Acoustic Phase Distortions in a Motor Vehicle

The method of the invention uses the difference between the acoustic distortions at the left-hand and right-hand front locations mainly in the frequency bands for which said distortions have different signs.

System for generating immersive audio utilizing visual cues

The present disclosure is directed to a system for generating immersive audio utilizing visual cues. In general, a system may be capable of transforming non-spatial audio (e.g., simple mono or stereo sound) associated with video into immersive sound (e.g., wherein sound may be generated spatially so that it appears to emanate directly from sound sources in the video). An example device may comprise data sourcing circuitry to receive multimedia data including at least video and non-spatial audio corresponding to the video, data analysis circuitry to determine at least one source of audio in the video and at least one attributable sound in the audio and audio generation circuitry to generate immersive audio wherein the at least one attributable sound is at least spatially associated with the at least one source of audio. Audio sources in the video may be determined using a variety of different types of detection.

Audio Signal Processing Apparatus

The disclosure is based on the finding that acoustic near-field transfer functions indicating acoustic near-field propagation channels between loudspeakers and ears of a listener can be employed to pre-process audio signals. Therefore, acoustic near-field distortions of the audio signals can be mitigated. The pre-processed audio signals can be presented to the listener using a wearable frame, wherein the wearable frame comprises the loudspeakers for audio presentation. The disclosure can allow for a high quality rendering of audio signals as well as a high listening comfort for the listener. The disclosure can provide the following advantages. By means of a loudspeaker selection as a function of a spatial audio source direction, cues related to the listener's ears can be generated, making the approach more robust with regard to front/back confusion. The approach can further be extended to an arbitrary number of loudspeaker pairs. 1. An audio signal processing apparatus comprising: provide a first acoustic near-field transfer function of a first acoustic near-field propagation channel between a first loudspeaker and a left ear of a listener; and', 'provide a second acoustic near-field transfer function of a second acoustic near-field propagation channel between a second loudspeaker and a right ear of the listener; and, 'a provider configured to filter a first input audio signal based on a first inverse of the first acoustic near-field transfer function to obtain a first output audio signal that is independent of a second input audio signal; and', 'filter the second input audio signal based on a second inverse of the second acoustic near-field transfer function to obtain a second output audio signal that is independent of the first input audio signal., 'a filter coupled to the provider and configured to2. The audio signal processing apparatus of claim 1 , further comprising a memory for providing the first acoustic near-field transfer function and the second acoustic ...

APPARATUS HAVING AN INPUT AND AN OUTPUT AND HAVING AN EFFECT DEVICE WITH VOLUME-REGULATED AUDIO SIGNALS FROM AN AUDIO FILE

The invention relates to an apparatus () having an input () and an output () and having an effect device with volume-regulated audio signals from an audio file or audio source, wherein the audio signals have input levels at an input of the effect device and output levels at an output of the effect device. In order to achieve background sound which is as optimal as possible, the invention proposes that each input level is assigned to a preset output level irrespective of the content of the audio file or audio source, wherein the input level is regulated to the output level. 1. A device , comprisingat least one input,at least one output, andan effects unit with volume-regulated audio signals of an audio file or audio source,wherein the audio signals have input levels at an input of the effects unit and output levels at an output of the effects unit,wherein irrespective of the content of the audio file or audio source, each input level is assigned to a preset output level,wherein the input level is regulated to the output level.2. The device as claimed in claim 1 , wherein the effects unit is an audio compressor.3. The device as claimed in claim 1 , wherein the output level is a value of a non-linear characteristic curve obtained by measured values.4. The device as claimed in claim 1 , wherein the output level is a value of a linear characteristic curve obtained by measured values.5. The device as claimed in claim 1 , further comprising a leveler arranged between the at least one input and the effects unit claim 1 , wherein the leveler is connected to the at least one input and the effects unit.6. The device as claimed in claim 1 , further comprising a limiter is arranged between the at least one output and the effects unit claim 1 , wherein the limiter is connected to the at least one output and the effects unit.7. The device as claimed in claim 6 , further comprising a spatializer arranged between the at least one output and the limiter claim 6 , wherein the ...

Crosstalk cancellation for opposite-facing transaural loudspeaker systems

Embodiments relate to audio processing for opposite facing speaker configurations that results in multiple optimal listening regions around the speakers. A system includes a left speaker and a right speaker in an opposite facing speaker configuration, and a crosstalk cancellation processor connected with the left speaker and the right speaker. The crosstalk cancellation processor applies a crosstalk cancellation to an input audio signal to generate left and right output channels. The left output channel is provided to the left speaker and the right output channel is provided to the right speaker to generate sound including multiple crosstalk cancelled listening regions that are spaced apart.

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 , ...

Binaural beat sound output device having improved sound field and method therefor

The present invention relates to a binaural beat sound output device having an improved sound field and a method therefor and, more specifically, to: a brainwave sound output device using binaural beats, which receives music desired by a user and applies binaural beats to a specific frequency band thereof, so that the device can provide the music through a stereo sound source while inducing a desired brainwave state by the stereo sound source, and thus can improve the sound field without the sense of rejection or monotony to the music; and a method therefor.

METHOD FOR GENERATING BINAURAL SIGNALS FROM STEREO SIGNALS USING UPMIXING BINAURALIZATION, AND APPARATUS THEREFOR

Disclosed is an audio signal processing method including: receiving a stereo signal; transforming the stereo signal into a frequency-domain signal; rendering the first signal based on a first ipsilateral filter coefficient; generating a frontal ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a second ipsilateral filter coefficient; generating a side ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a contralateral filter coefficient; generating a side contralateral signal relating to the frequency-domain signal; transforming an ipsilateral signal, generated by mixing the frontal ipsilateral signal and the side ipsilateral signal, and the side contralateral signal into a time-domain ipsilateral signal and a time-domain contralateral signal, which are time-domain signals, respectively; and generating a binaural signal by mixing the time-domain ipsilateral signal and the time-domain contralateral signal. 1. An audio signal processing method comprising:receiving a stereo signal;transforming the stereo signal into a frequency-domain signal;separating the frequency-domain signal into a first signal and a second signal based on an inter-channel correlation and an inter-channel level difference (ICLD) of the frequency-domain signal, wherein the first signal includes a frontal component of the frequency-domain signal, and the second signal includes a side component of the frequency-domain signal;rendering the first signal based on a first ipsilateral filter coefficient and generating a frontal ipsilateral signal relating to the frequency-domain signal, wherein the first ipsilateral filter coefficient is generated based on an ipsilateral response signal of a first head-related impulse response (HRIR);rendering the second signal based on a second ipsilateral filter coefficient and generating a side ipsilateral signal relating to the frequency-domain signal, wherein the second ...

Representation of contextual information by projecting different participants' audio from different positions in a 3D soundscape

An overlay network platform facilitates a multi-party conference. End users participate in the conference using client-based web browser software. According to this disclosure, an enhanced “audio” experience for a user participating in the conference is provided by rendering different participants' audio in the conference at different positions in a three-dimensional (3D) space. In operation, and given a set of audio streams that comprise the conference and that are being received, a three-dimensional (3D) position for each audio stream in a 3D soundscape is computed. In one embodiment, the 3D position for each audio stream is computed by identifying a target area region of space, and then determining the 3D position for each stream in the target area region of space. The relative 3D positions may be determined based on contextual information related to the content of an audio stream, as well as a computed “relevance” of one stream versus another. The streams are then rendered to the listener based on the computed 3D positions. 1. A method carried out at a given client machine , the given client machine being one of a set of client machines participating in a multi-party conference using an overlay network platform , the multi-party conference comprising a set of audio streams received from multiple users each associated with a respective client machine , comprising:computing a position in a soundscape for each of the received audio streams by identifying a target area region of space in front of the user of the given client machine, and identifying a relative position of each of a set of remote users in the multi-party conference as follows: (a) applying a function to a unique identifier associated with the remote user of the set of remote users to generate a number, (b) based on the number generated for each of the set of remote users, assigning each of the remote users a relative left-right position within the target area region of space in front of the user; ...

NARROW PROFILE SPEAKER CONFIGURATIONS AND SYSTEMS

A narrow profile speaker unit comprises at least one speaker outputting sound towards an internal surface and through a duct with an output terminus, such as a slot, having a narrow dimension, effectively changing the cross-section of the speaker's audio output wave. A pair of speakers may face one another, outputting sound towards a common output slot. Multiple pairs of speakers may be used to form an inline speaker unit for increased sound output. A slotted speaker unit may include multiple speakers facing the same direction, towards a groundplane or reflecting surface, and having parallel apertures for allowing sound radiation. The speaker units may be integral with or attached to electronic appliances such as desktop computers or flatscreen devices, or may be used in automobiles or other contexts. 1a speaker disposed on a mounting surface; anda sound reflecting surface disposed in front of the speaker face and substantially parallel with the mounting surface, the sound reflecting surface and the mounting surface collectively defining a sound duct terminating in an output slot, whereby the sound output from the speaker emanates from the output slot.. A narrow profile sound system, comprising: This application is a continuation of U.S. application Ser. No. 15/173,270 filed Jun. 3, 2016, which is a continuation of U.S. application Ser. No. 13/798,006 filed Mar. 12, 2013, which is a continuation of U.S. application Ser. No. 13/208,923 filed Aug. 12, 2011, which is a continuation of Ser. No. 12/246,433 filed Oct. 6, 2008 now U.S. Pat. No. 8,027,500, which is a divisional application of U.S. application Ser. No. 10/937,796 filed Sep. 8, 2004, now U.S. Pat. No. 7,433,488, which is a continuation-in-part application of U.S. application Ser. No. 10/339,357 filed Jan. 8, 2003, now U.S. Pat. No. 7,457,425, which is a continuation-in-part application of utility application U.S. application Ser. No. 10/074,604 filed on Feb. 11, 2002, now U.S. Pat. No. 7,254,239, (which ...

CAR STEREO AUDIO EXPANDING METHOD

The present invention relates to the technical field of a car stereo audio expanding method. The method performs delay processing on sound channel signals in a loudspeaker twice to counteract signals reflected by door glass, such that the sound field performs better. The sound field processing method of the present invention can enable the sound field of sounds heard by the driver to be centered and the diffusion of both sides to be symmetric. Also the sound field can be expanded to the outside of the vehicle, resolving the problem that the car audio, in a car environment, has a sound field weighted toward one side, has a narrow sound field width, and has poor hearing effects due to the confined sound field. The invention enables sound heard to be authentic rather than empty. 1: A car stereo audio expanding method , comprising steps of:{'b': '10', 'S, performing different degrees of first delay processing on a first sound channel signal output to a first loudspeaker near a driving seat and a second sound channel signal output to a second loudspeaker far from the driving seat according to a relative position;'}{'b': '20', 'S, outputting the first sound channel signal processed by the first delay to a first sound mixer, and performing a second delay processing on the first sound channel signal according to a relative position and outputting to a second sound mixer; outputting the second sound channel signal processed by the first delay to the second sound mixer, and performing the second delay processing on the second sound channel signal according to a relative position and outputting to the first sound mixer;'}{'b': '30', 'S, after performing reverse operation to the second sound channel signal processed by the second delay, adding it to the first sound channel signal output to the first sound mixer to obtain a sound channel signal for driving the first loudspeaker; after performing reverse operation to the first sound channel signal processed by the second delay, ...

Audio Generation in a Media Playback System

Example techniques relate to audio generation in a media playback system. Based on one or more first functions and first characteristics of an area, the system may generate first audio that includes a first audio signal and a second audio signal. The system provides the first audio signal to at least one first audio driver and the second audio signal to at least one second audio driver, thereby causing a first playback device and a second playback device to play back the first audio synchronously. The system receives second characteristics of the area and based on one or more second functions and the second characteristics, generates second audio comprising a third audio signal and a fourth audio signal. The system provides the third audio signal to the at least one first audio driver and the fourth audio signal to the at least one second audio driver. 1. A media playback system comprising multiple processors and multiple audio drivers carried by multiple playback devices , wherein a first playback device of the multiple playback devices comprises: a first network interface; a first audio amplifier configured to drive at least one first audio driver of the multiple audio drivers; and a first processor of the multiple processors , wherein a second playback device of the multiple playback devices comprises: a second network interface; a second audio amplifier configured to drive at least one second audio driver of the multiple audio drivers; and a second processor of the multiple processors , and wherein the media playback system further comprises data storage and instructions stored in the data storage , wherein the instructions , when executed by at least one processor of the multiple processors , cause the media playback system to perform functions comprising:based on one or more first functions and one or more first characteristics of an area, generating first audio comprising a first audio signal and a second audio signal;providing the first audio signal to the ...

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 SIGNAL PROCESSING METHOD AND DEVICE

A method and apparatus for audio signal processing in an audio chain to correct a non-linearity of the electroacoustic transducers in the audio chain by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain using an approximation of the quadratic function. The method accommodates the psychoacoustical characteristics of the human ear by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain approximating by a non-linear fifth degree polynomial function for a pressure change by the human ear up to p_Δ. The method and apparatus reduce non-linearities of the entire audio chain with the human ear, by adding non-linearities in the audio chain so that an audio chain characteristic reduces the non-linearity of the human ear polynomial approximation to the pressure change p_Δ=±1 Pa.

Apparatus with an audio equalizer and associated method

An apparatus comprising: a processor and memory including computer program code, the memory and the computer program code configured to, with the processor, cause the apparatus to: display a user interface for an audio equalizer, the user interface comprising a predefined region associated with a particular category of audio equalization; detect the position of a user input within the predefined region; and control the relative amplitudes of one or more frequency bandwidths of an audio signal to be output based on the detected position within the predefined region to produce a specific audio output having audio characteristics associated with that particular audio equalization category.

Subband Spatial and Crosstalk Cancellation for Audio Reproduction

Embodiments herein are primarily described in the context of a system, a method, and a non-transitory computer readable medium for producing a sound with enhanced spatial detectability and reduced crosstalk interference. The audio processing system receives an input audio signal, and performs an audio processing on the input audio signal to generate an output audio signal. In one aspect of the disclosed embodiments, the audio processing system divides the input audio signal into different frequency bands, and enhances a spatial component of the input audio signal with respect to a nonspatial component of the input audio signal for each frequency band.

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.

MULTI-MODE VIDEO CONFERENCING SYSTEM

The present disclosure is directed to a video-enabled communication system that comprises a control unit, coupled with a camera, the camera acquiring an image of a local participant to provide to a remote participant during a video communication session. A microprocessor in the control unit executes an audio controller that causes the microprocessor to control, based on audio information collected by the control unit, movement and/or pan and/or tilt of a camera at a remote endpoint of the remote participant to capture a selected image of the remote participant for display to the local participant. The control unit can operate in multiple operating modes depending on the context of the video conference. 1. A video-enabled communication system , comprising:a microprocessor, coupled with a camera, to receive an image of an object of interest during a video communication session; anda computer readable medium comprising instructions that program the microprocessor: (a) in a first operating mode, use a first set of instructions to perform a first video communication operation or function and (b) in a second operating mode, use a different set of instructions to perform the first video communication operation or function; andan imaging controller that causes the microprocessor to move the camera from a first spatial location to a second spatial location relative to a selected point of reference to enable the camera to capture different images of a local participant for display, by a remote display device, to a remote participant to the video communication session; andwherein the camera is a pan, tilt, and zoom camera, wherein, as a result of the movement, the pan rotational axis moves from the first spatial location to the second spatial location, and wherein the camera movement moves the camera linearly or curvilinearly from the first to the second spatial locations; and a carriage displacement unit to cause camera movement along the at least one track; and', 'a ...

Method and apparatus for acoustic crosstalk cancellation

A crosstalk canceller for reducing acoustic crosstalk at a time of audio playback is derived by forming a channel frequency response for a nominated playback geometry, and decomposing the channel frequency response to derive a decomposition element such as a singular value decomposition matrix. A value of the decomposition element, such as the smallest singular value, is then adjusted to reduce spectral coloration, and filter coefficients of a crosstalk cancellation filter are derived from the adjusted decomposition element.

METHOD AND DEVICE FOR VOICE OPERATED CONTROL

Methods and devices for processing and voice operated control are provided and mixing ambient and internal signals. The method can include performing a non-difference comparison between a first received sound and a second received sound, determining if speech exists based on the comparison, and transmitting or providing a decision that the speech is present to at least one among the device, a cell phone, a media player, or a portable computing device. Other embodiments are disclosed. 1. An earpiece , comprising:a sealing element, wherein the sealing element is configured to provide a passive level of sound reduction between a first side of the earpiece and a second side of the earpiece when the earpiece is worn by a user;an ambient microphone (ASM) configured to measure sound on the first side of the earpiece to generate an ambient signal;a speaker configured to deliver acoustic energy into the ear canal of the user;an ear canal microphone (ECM) configured to measure sound on the second side of the earpiece to generate an internal signal;a memory that stores instructions; and receiving the ambient signal from the ambient microphone;', 'receiving the internal signal from the ear canal microphone;', 'sending an audio signal to the speaker;', 'analyzing at least one of the ambient signal, the internal signal, or a combination thereof, to detect a voice; and', 'stopping the sending of the audio signal to the speaker or send a modified audio signal to the speaker, if a voice is detected., 'a processor operatively coupled to the ambient microphone, the ear canal microphone, the memory, and the speaker, wherein the processor is configured to execute the instructions to perform operations, the operations comprising2. An earpiece according to claim 1 , wherein the operation of analyzing uses at least one of a correlation claim 1 , cross-correlation claim 1 , level detection claim 1 , spectral analysis claim 1 , coherence claim 1 , peak detection claim 1 , signal ratio or a ...

ENTERTAINMENT AUDIO PROCESSING

An entertainment audio processing system includes a music limiter configured to receive an audio signal, volume control information for music, and volume control information for announcement. The music limiter is further configured to calculate a gain parameter based on a music-to-announcement ratio. In another aspect, an entertainment audio processing system includes a mixer configured to substantially match a left audio signal channel and a right audio signal channel. The entertainment audio processing system further includes an equalizer configured to introduce magnitude and phase changes between the left audio signal channel and the right audio signal channel. 1. An apparatus comprising:a mixer configured to at least partially collapse an audio signal to mono by making a left audio signal channel and a right audio signal channel similar;an equalizer coupled to an output of the mixer, the equalizer configured to introduce magnitude and phase differences between the left audio signal channel and the right audio signal channel to generate a processed audio signal;a latency compensator configured to receive an unprocessed audio signal and introduce a time delay that matches a time delay associated with the processed audio signal to generate a time-delayed audio signal; andan x-fade coupled to an output of the equalizer and an output of the latency compensator, the x-fade configured to output a digital audio signal by selecting to output one of the processed audio signal or the time-delayed audio signal based, at least in part, upon an announcement trigger signal received from a non-entertainment audio processing system.2. The apparatus of claim 1 , wherein:the announcement trigger signal provides an indication that an announcement audio is present.3. The apparatus of claim 1 , where in the time delay associated with the processed audio signal is due to at least one of the mixer and the equalizer.4. The apparatus of claim 1 , wherein the magnitude and phase ...

Spatial Crosstalk Processing for Stereo Signal

An audio system provides for crosstalk processing and crosstalk compensation processing of an audio signal. The crosstalk processing may include crosstalk cancellation processing or crosstalk simulation processing. A crosstalk processed signal is generated by applying the crosstalk processing to a side channel of the left and right channels, with a mid channel of the left and right channels bypassing the crosstalk processing. The crosstalk processed signal and the mid channel that bypasses crosstalk processing is used to generate a left output channel and a right output channel. In some embodiments, a crosstalk compensated signal is generated by applying crosstalk compensation processing to the side channel. The crosstalk compensated signal adjusts for spectral defects caused by the crosstalk processing. The crosstalk processing and crosstalk compensation processing may be applied in different orders. The left and right output channels are generated using the crosstalk processed signal and the crosstalk compensated signal. 1. A method for enhancing an audio signal having a left channel and a right channel , the method comprising:applying crosstalk processing including a filter and a delay to a side channel of the left channel and the right channel to generate a crosstalk processed signal, the side channel including a difference between the left channel and the right channel, and a mid channel of the left channel and the right channel bypassing the crosstalk processing, the mid channel including a sum of the left channel and the right channel; andgenerating a left output channel and a right output channel using the crosstalk processed signal and the mid channel that bypasses the crosstalk processing.2. The method of claim 1 , wherein the crosstalk processing includes one of a crosstalk cancellation processing or a crosstalk simulation processing.3. The method of claim 1 , wherein:the crosstalk processing includes a crosstalk cancellation processing; and separating ...

Method and apparatus for decoding stereo loudspeaker signals from a higher-order ambisonics audio signal

Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localisation in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA.

Method and an apparatus for generating an acoustic signal with an enhanced spatial effect

An apparatus and a method for generating an acoustic signal with an enhanced spatial effect, said apparatus comprising a signal filter bank adapted to filter a difference audio signal with a filter characteristic to limit a bandwidth of said difference audio signal, wherein said bandwidth limited difference audio signal is applied to at least one pair of loudspeakers for dipole sound emission.

TINNITUS TREATMENT SYSTEM AND METHOD

A tinnitus masking system for use by a person having tinnitus. The system comprises a sound delivery system having left and right ear-level audio delivery devices and is configured to deliver a masking sound to the person via the audio delivery devices such that the masking sound appears to originate from a virtual sound source location that substantially corresponds to the spatial location in 3D auditory space of the source of the tinnitus as perceived by the person. The masking sound being represented by left and right audio signals that are converted to audible sound by the respective audio delivery devices. 1. A system for determining a spatial property of tinnitus as perceived by a person comprising:a sound generation system that is operable to present test sounds to the person from a range of virtual sound source locations in 3D auditory space; and receive person feedback indicative of the virtual sound source location that most closely corresponds to the spatial location in 3D auditory space of the source of the tinnitus as perceived by the person and', "output spatial information indicative of the spatial location of the source of the tinnitus based on the person's feedback."], 'a feedback system that is arranged to'}2. A system according to claim 1 , wherein the sound generation system is configured to sequentially present test sound to the person from a range of different virtual sound source locations.3. A system according to claim 1 , wherein the sound generation system is user operable to present the test sounds from user selected virtual sound source locations.4. A system according to claim 1 , wherein the test sound has one or more sound attributes that substantially correspond or match one or more of the perceived sound attributes of the person's tinnitus.5. A system according to claim 4 , wherein the sound attributes comprise any one or more of the following: pitch claim 4 , frequency claim 4 , bandwidth claim 4 , temporal properties claim 4 , ...

Three-Dimensional Sound Human Machine Interface, Navigation, And Warning Systems In A Vehicle

Systems and methods for providing a vehicle occupant with a three-dimensional sound human machine interface, three-dimensional sound navigation, and three-dimensional sound warnings. One method includes retrieving a menu option, determining a virtual location of the menu option relative to the vehicle occupant, and emitting audio related to the menu option through a three-dimensional sound system to cause the vehicle occupant to hear the audio at the virtual location. The method also includes receiving feedback from the vehicle occupant related to the menu option and interpreting the feedback to adjust the virtual location of the menu option. 1. A method for providing a vehicle occupant with a three-dimensional sound human machine interface , the method comprising:retrieving a menu option;determining a virtual location of the menu option relative to the vehicle occupant;emitting audio related to the menu option through a three-dimensional sound system to cause the vehicle occupant to hear the audio from the virtual location;receiving feedback from the vehicle occupant relating to the menu option; andinterpreting the feedback to adjust the virtual location of the menu option.2. The method of claim 1 , comprising interpreting the feedback to select the menu option.3. The method of claim 2 , wherein the feedback includes one of a swipe gesture and a select gesture.4. The method of claim 2 , wherein selecting the menu option includes emitting media system audio related to the menu option through the three-dimensional sound system throughout the vehicle.5. The method of claim 2 , wherein selecting the menu option includes retrieving a new menu option.6. The method of claim 1 , comprising recording the feedback through a depth camera.7. The method of claim 1 , comprising interpreting the feedback to adjust a volume of the audio.8. The method of claim 1 , comprising retrieving a second menu option;determining a second virtual location of the second menu option relative to ...

Active channel crossover system

The contained Pass And Limit Modulator removes approximately 50% common or in phase information from two inputs as compared to summing both inputs. This output can be used in a whole building music system or in a mono amplitude modulated radio transmission down-mixed from stereo audio content. The center channel (dialogue) is lowered 50% as not to overpower the music. With the Active Channel Crossover System the center channel is further processed and mixed to obtain multiple output channels. Interactive left and right channel frequency decomposition of a stereo signal is performed for optimum reproduction of a down converted front center channel. This way channel removal speed is respective to the frequencies removed. Further processing reduces the interactive left and right channel frequency decomposed content within the center channel output. As a result, the down-mixed front center channel is reproduced more closely as intended to be heard in its original stereo form. Further processing within the Active Channel Crossover System produces a balanced multi channel music system.

Audio processor for orientation-dependent processing

Audio processor having an input interface, a detector interface, a mixer and an output interface. The input interface receiving at least two input audio channels, each input audio channel being associated with a predetermined reproduction position of at least two loudspeakers on at least one loudspeaker axis. The detector interface receiving a position signal indicating an information on a position of the at least two loudspeakers with respect to an ear axis of a listener, wherein the ear axis and the at least one loudspeaker axis have an angle to each other, being greater than 0° and lower than 180°. The mixer mixing the at least two input audio channels to obtain the at least two output channels depending on the position signal.

SYSTEM AND METHOD FOR GENERATING SPATIAL SOUND USING ULTRASOUND

A novel system and method for spatial sound generation is disclosed. A system and method for generating bodiless mid-air speakers includes the steps of: generating a modulated signal by modulating an ultrasonic carrier signal with an audio signal, determining a phase delay value for each ultrasonic transducer of an array of ultrasonic transducers with respect to one or more focal points, and driving each such ultrasonic transducer with the modulated signal in accordance with the phase delay value determined for each ultrasonic transducer to generate audible sound at the one or more focal points. 120-. (canceled)21. A method for generating bodiless mid-air speakers comprising the steps of:generating a modulated signal by modulating an ultrasonic carrier signal with an audio signal;calculating a time delay for the application of a driving signal to each ultrasonic transducer of an array of ultrasonic transducers with respect to one or more focal points; andapplying the driving signal to each ultrasonic transducer of the array of ultrasonic transducers with the modulated signal in accordance with the time delay calculated for the ultrasonic transducer to focus ultrasound at the one or more focal points.22. The method of claim 21 , wherein the time delay for each ultrasonic transducer is calculated based on the distance between the ultrasonic transducer and the focal point.23. The method of claim 22 , wherein the time delay for each ultrasonic transducer is calculated in accordance with the equation Δtij=−lij/c claim 22 , wherein Δtij represents the time delay claim 22 , lij represents the distance from the focal point to the ultrasonic transducer (i claim 22 ,j) claim 22 , and c represents a speed of sound.24. The method of claim 21 , wherein the time delay is calculated to set one or more focal points near an object or image so as to make it appear as if the object or image is the source of audible sound.25. The method of comprising the additional steps of: ...

METHOD FOR OUTPUTTING AN AUDIO SIGNAL INTO AN INTERIOR VIA AN OUTPUT DEVICE COMPRISING A LEFT AND A RIGHT OUTPUT CHANNEL

Method for outputting an audio signal (S) into an interior () forming part of a passenger compartment () of a motor vehicle () via an acoustic output device () comprising a left and a right acoustic output channel (), comprising the steps: 1423756. Method for outputting an audio signal (S) into an interior () forming part of a passenger compartment () of a motor vehicle () via an acoustic output device () comprising a left and a right acoustic output channel ( , ) , comprising the steps:supplying an audio signal (S),{'b': 1', '3, 'splitting of the supplied audio signal (S) into a plurality of acoustic audio signal components (S.—S.) dependent on the direction of perception,'}{'b': 1', '3', '5', '6', '7', '5', '6, 'mixing of each acoustic audio signal component (S.—S.) dependent on the direction of perception to a right and/or left output channel (, ) of an acoustic output device () comprising a right and a left acoustic output channel (, ),'}{'b': 1', '3', '5', '6', '7, 'output of the acoustic audio signal components (S.—S.) dependent on the direction of perception via the right and/or the left acoustic output channel (, ) of the acoustic output device ().'}221232. Method according to claim 1 , characterized in that the supplied audio signal (S) is split into a center audio signal component (S.) claim 1 , a left audio signal component (S.) perceived to the left of the center audio signal component (S.) and a right audio signal component (S.) perceived to the right of the center audio signal component (S.).31573672567. Method according to claim 2 , characterized in that the left audio signal component (S.) is output via the left acoustic output channel () of the acoustic output device () claim 2 , the right audio signal component (S.) is output via the right acoustic output channel () of the acoustic output device () claim 2 , and the center audio signal component (S.) is output via the left and/or via the right output channel ( claim 2 , ) of the output device (). ...

Efficient personalization of head-related transfer functions for improved virtual spatial audio

A method generating a virtual audio signal for a listener. The method includes estimating spherical harmonic coefficients based on an individual character of the listener. The estimated spherical harmonic coefficients are compared to a distribution of known spherical harmonic coefficients. The estimated spherical harmonic coefficients are iteratively updated and compared to the distribution of known spherical harmonic coefficients until convergence. The individual character and the converged spherical harmonic coefficients are then applied to a mono-channel sound.

BINAURAL RECORDING SYSTEM AND EARPIECE SET

Systems, methods, and sets for capturing and reproducing a binaural recording are provided. In one embodiment, the binaural recording is captured using an earpiece set that includes a pair of earpieces each including a frame, a microphone, and a transducer. The frame is adapted to be worn on an ear, and defines an ear-insertion end and a posterior end opposite the ear-insertion end. The ear-insertion end is adapted to be positioned within an ear canal. The microphone is coupled to the frame for capturing ambient sound, and adapted to be positioned outside the ear canal and substantially coplanar with a tragus of the ear. 1. A binaural recording system for recording a sound from an impulsive source comprising: a right frame, the right frame adapted to be worn on a right ear, the right frame defining a right ear-insertion end and a right posterior end opposite the right ear-insertion end, the right ear-insertion end adapted to be positioned within a right ear canal,', 'a right microphone, the right microphone coupled to the right frame and the right microphone adapted to be positioned outside the right ear canal and substantially coplanar with a right tragus of the right ear, wherein the right microphone captures a right side frequency from the impulsive sound source, and', 'a right transducer, the right transducer coupled to the right ear-insertion end of the right frame, and the right transducer configured to reproduce a right sound;, 'a right earpiece, the right earpiece including a left frame, the left frame adapted to be worn on a left ear, the left frame defining a left ear-insertion end and a left posterior end opposite the left ear-insertion end, the left ear-insertion end adapted to be positioned within a left ear canal,', 'a left microphone, the left microphone coupled to the left frame and the left microphone adapted to be positioned outside the left ear canal and substantially coplanar with a left tragus of the left ear, wherein the left microphone ...

WIND NOISE REDUCTION

A method of wind noise reduction. Left side and right side microphone signals are obtained. In a first stage wind noise reduction is applied to a first sub-band of one of the signals, below a spectral threshold N. In a second stage the wind noise reduced first side signal, and the second side signal, are both split into a sub-band below a threshold Nless than N. The sub-band of the first side signal is mixed with the sub-band of the second side signal to produce an aggregate third sub-band signal having reduced wind noise, which is recombined with the respective sub-bands above the threshold Nto produce output first and second side signals. 1. A method of wind noise reduction , the method comprising:deriving from a plurality of microphones at least one first side input signal and at least one second side input signal, the first and second sides each being one of a left side and a right side; [{'sub': A', 'A, 'splitting the first side signal into a first sub-band below a spectral threshold Nand a second sub-band above the spectral threshold N;'}, 'applying wind noise reduction to the first sub-band of the first signal to produce a wind noise reduced first sub-band of the first signal; and', 'recombining the wind noise reduced first sub-band of the first signal with the second sub-band of the first signal, to produce a wind noise reduced first side signal;, 'in a first stage [{'sub': B', 'B, 'splitting the wind noise reduced first side signal into a third sub-band below a spectral threshold Nand a fourth sub-band above the spectral threshold N;'}, {'sub': B', 'B, 'splitting the second side signal into a third sub-band below the spectral threshold Nand a fourth sub-band above the spectral threshold N;'}, 'mixing the third sub-band of the first side signal with the third sub-band of the second side signal to produce an aggregate third sub-band signal having reduced wind noise;', 'combining the aggregate third sub-band signal with the fourth sub-band of the first side ...

Sound Localization for an Electronic Call

During an electronic call between two individuals, a sound localization point simulates a location in empty space from where an origin of a voice of one individual occurs for the other individual. 1. A method , comprising:capturing, with an electronic earphone located at a head of a talking person, binaural sound that will be provided to a listening person during a telephone call;designating, with a computer system, a sound localization point in empty space that is away from and proximate to the listening person such that the sound localization point simulates an origin of the binaural sound at the empty space that the listening person hears during the telephone call;adjusting, with the computer system, the binaural sound captured at the earphone of the talking person so the binaural sound originates during the telephone call from the sound localization point in empty space that is away from and proximate to the listening person; andproviding, with an electronic earphone located at a head of the listening person, the binaural sound to the listening person during the telephone call such that the origin of the binaural sound for the listening person occurs at the sound localization point in empty space that is away from and proximate to the listening person.2. The method of further comprising:tracking a head orientation of the listening person with respect to the sound localization point;adjusting an interaural time difference of the sound that the listening person hears in response to the listening person changing the head orientation with respect to the sound localization point.3. The method of further comprising:receiving, from the listening person, a location that specifies a distance from the listening person where to position the sound localization point in empty space that is proximate to and away from the listening person.4. The method of further comprising:receiving, from the listening person, a designation of an object that is not an electronic device and ...

Methods and apparatus for implementing sound events

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

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; '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', '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, whereinoutputting 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 , wherein the coordinate ...

Sound Localization for an Electronic Call

During an electronic call between two individuals, a sound localization point simulates a location in empty space from where an origin of a voice of one individual occurs for the other individual. 1. A method , comprising:capturing, with an electronic earphone located at a head of a talking person, binaural sound that will be provided to a listening person during a telephone call;designating, with a computer system, a sound localization point in empty space that is away from and proximate to the listening person such that the sound localization point simulates an origin of the binaural sound at the empty space that the listening person hears during the telephone call;adjusting, with the computer system, the binaural sound captured at the earphone of the talking person so the binaural sound originates during the telephone call from the sound localization point in empty space that is away from and proximate to the listening person; andproviding, with an electronic earphone located at a head of the listening person, the binaural sound to the listening person during the telephone call such that the origin of the binaural sound for the listening person occurs at the sound localization point in empty space that is away from and proximate to the listening person.2. The method of further comprising:tracking a head orientation of the listening person with respect to the sound localization point;adjusting an interaural time difference of the sound that the listening person hears in response to the listening person changing the head orientation with respect to the sound localization point.3. The method of further comprising:receiving, from the listening person, a location that specifies a distance from the listening person where to position the sound localization point in empty space that is proximate to and away from the listening person.4. The method of further comprising:receiving, from the listening person, a designation of an object that is not an electronic device and ...

Methods, apparatus and systems for asymmetric speaker processing

A method of processing audio data for replay on a mobile device with a first speaker and a second speaker, wherein the audio data comprises a respective audio signal for each of the first and second speakers, includes: determining a device orientation of the mobile device; if the determined device orientation is vertical orientation, applying a first processing mode to the audio signals for the first and second speakers; and if the determined device orientation is horizontal orientation, applying a second processing mode to the audio signals for the first and second speakers. Applying the first processing mode involves: determining respective mono audio signals in at least two frequency bands based on the audio signals for the first and second speakers; in a first one of the at least two frequency bands, routing a larger portion of the respective mono audio signal to one of the first and second speakers; and in a second one of the at least two frequency bands, routing a larger portion of the respective mono audio signal to the other one of the first and second speakers. Applying the second processing mode involves applying cross-talk cancellation to the audio signals for the first and second speakers.

VIRTUAL SOUNDSTAGE WITH COMPACT SPEAKER ARRAY AND INTERAURAL CROSSTALK CANCELLATION

A system and method for generating a virtual soundstage in a listening environment having a compact speaker array centrally positioned in a listening environment. A listener is sitting offset from the speaker array and the virtual soundstage is generated in front of a listener. A signal processing unit is configured to receive an incoming audio signal, to process left and right channel signals of the incoming audio signal to generate a null, and to steer the null toward one ear of a listener thereby generating a virtual sound source that is offset from the center of the listening environment. Virtual sound sources are generated in front of, to the left of, and to the right of the listener. 1. A system for generating a virtual soundstage in a listening environment , the system comprising:a compact speaker array centrally positioned in a listening environment in front of a listener, a center of the compact speaker array coincides with a center of the listening environment, the compact speaker array has at least first and second speakers; anda signal processing unit configured to receive an incoming audio signal, to process left and right channel signals of the incoming audio signal to generate a null, to steer the null toward one ear of a listener thereby generating a virtual sound source that is offset from the center of the listening environment and directly in front of the listener.2. The system of claim 1 , wherein the signal processing unit is configured to feed an inverted signal to one of the speakers in the compact speaker array to generate the null.3. The system of claim 1 , wherein the signal processing unit is configured to introduce a predetermined time delay to the audio signal being played at the first or second speaker in the compact speaker array claim 1 , the predetermined time delay is introduced in the audio signal being played at the speaker in the compact speaker array that is closest to the one ear of the listener to steer the null toward the one ...

Electronic handheld audio/video receiver and listening/viewing device

A handheld device in an audio and video system including a system for receiving and processing video and/or audio signals, a system for displaying a virtual image to a user, and a system for producing sounds audible to a user. The handheld device also incorporates a shroud to block ambient light when the handheld unit is engaged with the user's face. As a result, the user may hear sounds produced from the received audio signals. Further the user may see the virtual images produced from the received video signals when the handheld device is engaged with the users face.

SPATIAL AUDIO ENHANCEMENT PROCESSING METHOD AND APPARATUS

An audio processing system for processing a single channel audio signal includes a a processor configured to derive a synthetic difference component from the single channel audio input signal a filtering module configured to apply a first filter to the sum signal represented by the single channel signal and to apply a second filter to the synthetic difference signal; and a control module configured to crossfade to control the amount of the resulting audio signal effect by respectively scaling the sum signal and the difference signal. 1. A method for upmixing a 2 channel audio signal comprising:generating a sum signal and a difference signal from the audio signal;applying a first filter to the sum signal; andapplying a second filter to the difference signal; wherein the gains on the first and the second filters are tuned to redistribute the mid and side contributions from the stereo input across the 2N output channels.2. The method as recited in wherein the gains are selected to satisfy a predetermined energy preservation characteristic of the implementing circuit.3. The method as recited in further comprising controlling the front-back energy distribution of the sum (M) and/or difference (S) components through user provided controls.4. The method as recited in further comprising decorrelating the back channels (B) relative to the front channels (F).5. The method as recited in further comprising combining the upmix gains and virtualization filters.6. The methods as recited in further comprising reducing the width of the frontal audio image by setting a sum fader to the front and a difference fader to the rear.7. The method as recited in further comprising applying early reflections to the virtual loudspeaker rendering provided by the 2N output channels and tuning the gains provided on the first and second filters to tune the appropriate balance of mid versus side components.8. An audio processing system for processing a single channel audio signal claim 1 , ...

PERSONALIZED HRTFS VIA OPTICAL CAPTURE

An apparatus and method of generating personalized HRTFs. The system is prepared by calculating a model for HRTFs described as the relationship between a finite example set of input data, namely anthropometric measures and demographic information for a set of individuals, and a corresponding set of output data, namely HRTFs numerically simulated using a high-resolution database of 3D scans of the same set of individuals. At the time of use, the system queries the user for their demographic information, and then from a series of images of the user, the system detects and measures various anthropometric characteristics. The system then applies the prepared model to the anthropometric and demographic data as part of generating a personalized HRTF. In this manner, the personalized HRTF can be generated with more convenience than by performing a high-resolution scan or an acoustic measurement of the user, and with less computational complexity than by numerically simulating their HRTF. 145-. (canceled)46. A method , comprising:generating an HRTF calculation system taking anthropometric measurements and, optionally, demographic data of a user as input and returning a personalized HRTF for the user as output,obtaining 3D scans of a plurality of training subjects,generating HRTFs for the plurality of training subjects by performing acoustic scattering calculations on anatomical characteristics captured in the 3D scans,collecting anthropometric measurements and, optionally, demographic data of the plurality of training subjects, andtraining the HRTF calculation system to transform the anthropometric measurements and, optionally, demographic data of the plurality of training subjects to the HRTFs.47. The method of claim 46 , wherein performing acoustic scattering calculations on anatomical characteristics captured in a 3D scan comprises performing a numerical simulation of the sound field around a mesh corresponding to the 3D scan.48. The method of claim 46 , wherein the ...

Representation of contextual information by projecting different participants' audio from different positions in a 3D soundscape

An overlay network platform facilitates a multi-party conference. End users participate in the conference using client-based web browser software. According to this disclosure, an enhanced “audio” experience for a user participating in the conference is provided by rendering different participants' audio in the conference at different positions in a three-dimensional (3D) space. In operation, and given a set of audio streams that comprise the conference and that are being received, a three-dimensional (3D) position for each audio stream in a 3D soundscape is computed. In one embodiment, the 3D position for each audio stream is computed by identifying a target area region of space, and then determining the 3D position for each stream in the target area region of space. The relative 3D positions may be determined based on contextual information related to the content of an audio stream, as well as a computed “relevance” of one stream versus another. The streams are then rendered to the listener based on the computed 3D positions. 1. A method carried out at a client machine , the client machine being one of a set of machines participating in a multi-party conference that is facilitated using an overlay network platform , comprising:given a set of audio streams that comprise the conference and that are being received, computing a three-dimensional (3D) position for each audio stream in a 3D soundscape; andrendering, in a browser, each audio stream at the 3D position computed.2. The method as described in wherein the browser is a web browser that renders the audio streams using a Hypertext Markup Language Version 5 (HTML5) Web Audio Application Programming Interface (API).3. The method as described in wherein the 3D position for each audio stream is computed by identifying a target area region of space claim 1 , and determining the 3D position for each stream in the target area region of space.4. The method as described in wherein claim 3 , with respect to a location of ...

Configurable multi-band compressor architecture with advanced surround processing

Audio processing architectures are described for implementing a multi-band audio compression algorithm with advanced surround processing. Exemplary architectures can improve the fidelity and perceived sound field spread of inexpensive, cabinet mounted, stereo speakers such as those that might be found in televisions, wireless speaker systems and sound bars. Embodiments of the present disclosure can improve inexpensive, cabinet mounted, stereo speakers by providing, e.g., (i) an Advanced Surround algorithm that adds depth and height to the left/right/center sound field images, (ii) a Soft Clip algorithm to minimize the perceived artifacts caused by compressor overshoot, (iii) configurable crossover filter order adjustment to allow better isolation between bands, (iv) a compressor maximum gain adjustment to reduce overshoot and minimize noise boost, and/or (v) a center gain adjustment to emphasize the perception of the center image (dialog) in high ambient noise situations.

Sound Localization for an Electronic Call

During an electronic call between two individuals, a sound localization point simulates a location in empty space from where an origin of a voice of one individual occurs for the other individual.

Method for outputting audio signal and electronic device supporting the same

A method and an electronic device for outputting an audio signal in the electronic device is provided. The electronic device includes a first speaker, a second speaker, and an audio processor that creates, from an audio signal, a first frequency audio signal corresponding to a first frequency band by using a low pass filter, synthesizes the created first frequency audio signal and the audio signal to create a synthetic audio signal, creates, from the synthetic audio signal, a second frequency audio signal corresponding to a second frequency band by using a high pass filter, outputs the created second frequency audio signal through the first speaker, and outputs the created synthetic audio signal through the second speaker.

Spatial Audio Signal Processing for Objects with Associated Audio Content

A user device for generating a scene, the user device comprising: an object determiner configured to determine an object for the scene, the object being associated with at least one audio signal; a relative position/orientation determiner configured to determine a relative position/orientation between the user device's user and the object; an audio position processor configured to spatially audio signal process the at least one audio signal to generate at least two channel audio signal based on the relative position/orientation. 1. A user device for generating a scene , the user device comprising:an object determiner configured to determine an object for the scene, the object being associated with at least one audio signal;a relative position/orientation determiner configured to determine a relative position/orientation between the user device's user and the object;an audio position processor configured to spatially audio signal process the at least one audio signal to generate at least two channel audio signal based on the relative position/orientation.2. The user device of claim 1 , wherein the audio position processor is configured to process the at least one audio signal with a head related transfer function with the relative position/orientation as a parameter input for the head related transfer function.3. The user device of claim 1 , wherein the scene is a shared scene communicated with at least one further user device over a communications network claim 1 , and wherein the object is associated with the at least one further user device claim 1 , and wherein the at least one audio signal is a transmitted audio signal from the at least one further user device.4. The user device of claim 1 , wherein the object is further associated with a video or image claim 1 , and the user device further comprise an object renderer configured to display the video or image at the relative position/orientation.5. The user device of claim 1 , wherein the relative position/ ...

Method and apparatus for acoustic crosstalk cancellation

An acoustic crosstalk canceller is determined for an asymmetric audio playback device, by determining a transfer function of an acoustic stereo playback path having asymmetries defined by speakers of the playback device. The transfer function is inverted to determine an inverse transfer function. The inverse transfer function is regularised by applying frequency dependent regularisation parameters to obtain an acoustic crosstalk canceller. Also, the inverse transfer function could be regularised for symmetric playback paths by applying aggregated frequency dependent regularisation parameters to obtain an acoustic crosstalk canceller without band branching.

MICROPHONE DEVICE TO PROVIDE AUDIO WITH SPATIAL CONTEXT

The disclosed technology generally relates to a microphone device configured to receive sound from different beams, where each beam has a different spatial orientation and is configured to receive sound from different directions. The microphone device is also configured to process the received sound using a generic or specific head related transfer function (HRTF) to generate processed audio and transmit the processed audio to hearing devices worn by a hearing-impaired user. Additionally, the microphone device can use a reference line and/or reference point when processing the received audio. 122-. (canceled)23. A microphone device comprising:a first and second microphone configured to individually or in combination form a sound receiving beam or beams;a processor electronically coupled to the first and second microphones, the processor configured to apply a digital signal processing algorithm to received sound at the sound receiving beam or beams based on an orientation of the sound receiving beam or beams based on a reference point to generate a multichannel output audio signal; anda transmitter configured to transmit the multichannel output audio signal generated by the processor, wherein the reference point is associated with a location on the microphone device.24. The microphone device of claim 23 , wherein the multichannel output audio signal is transmitted to hearing devices claim 23 , wherein a wearer of the hearing devices positioned the reference point relative to the wearer claim 23 , and wherein the reference point is associated with a virtual listener.25. The microphone device of claim 24 , wherein the processor weighs the received sound from a front claim 24 , left claim 24 , or right side of the virtual listener more than other received sound from the back of the virtual listener on the microphone device.26. The microphone device of claim 23 , wherein the multichannel output audio signal is a stereo signal.27. The microphone device of claim 23 , ...

AUDIO SIGNAL PROCESSING APPARATUS AND AUDIO SIGNAL PROCESSING METHOD

An audio signal processing apparatus includes a first bus, a second bus, and a controller that assigns a first input channel to the first bus, assigns a second input channel to the second bus, receives an amount of feed from the first input channel to the first bus, and receives an amount of feed from the second input channel to the second bus. 1. An audio signal processing apparatus comprising:a first bus;a second bus; anda controller that assigns a first input channel to the first bus, assigns a second input channel to the second bus, receives an amount of feed from the first input channel to the first bus, and receives an amount of feed from the second input channel to the second bus.2. The audio signal processing apparatus according to claim 1 , further comprising a signal processor that performs predetermined signal processing on an audio signal to be sent to the first bus.3. The audio signal processing apparatus according to claim 1 , wherein the first bus includes a signal processor that performs predetermined signal processing on an audio signal sent to the first bus.4. The audio signal processing apparatus according to claim 2 , wherein the predetermined signal processing includes processing that adds localization.5. The audio signal processing apparatus according to claim 1 , further comprising a plurality of physical operation elements claim 1 , wherein the controller receives the amount of feed from the first input channel to the first bus claim 1 , a gain setting for the first input channel claim 1 , the amount of feed from the second input channel to the second bus claim 1 , and a gain setting for the second input channel through the plurality of physical operation elements.6. The audio signal processing apparatus according to claim 5 , further comprising an operation panel on which the plurality of physical operation elements are arranged side-by-side claim 5 , wherein the controller assigns a physical operation element that receives the gain setting ...

Loudness adjustment for downmixed audio content

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

LOUDNESS ADJUSTMENT FOR DOWNMIXED AUDIO CONTENT

Disclosed is a non-transitory computer readable storage medium which receives, by an audio decoder (operating in a specific playback environment different from a reference channel configuration), an audio signal for the reference channel configuration. The audio signal includes audio sample data and encoder-generated loudness metadata which includes a plurality of portions of loudness metadata for a plurality of playback environments. The plurality of portions of loudness metadata includes one or more respective portions of loudness metadata for each playback environment in the plurality of playback environments. The medium also selects one or more portions of specific loudness metadata (based on the specific playback environment), from among the plurality of portions of loudness metadata for the plurality of playback environments. The one or more portions of specific loudness metadata relating to the specific playback environment determine loudness adjustment gains from the one or more portions of specific loudness metadata for the specific playback environment, apply the loudness adjustment gains as a part of overall gains applied to the audio sample data to generate output audio data. 1. A non-transitory computer readable storage medium , storing software instructions , which when executed by one or more processors cause performing:receiving, by an audio decoder operating in a specific playback environment different from a reference channel configuration, an audio signal for the reference channel configuration, the audio signal including audio sample data and encoder-generated loudness metadata, the encoder-generated loudness metadata comprising a plurality of portions of loudness metadata for a plurality of playback environments, the plurality of portions of loudness metadata comprising one or more respective portions of loudness metadata for each playback environment in the plurality of playback environments;selecting one or more portions of specific loudness ...

MULTI-CHANNEL MICROPHONE MAPPING

A method of adaptively mapping a plurality of microphone signals to a multi-channel audio signal, for example to capture stereo audio from a device held in variable orientations. At least first and second microphone signals, from respective first and second spaced apart microphones, are obtained. A device orientation signal is also obtained from a device orientation sensor. The microphone signals are adaptively mapped, based on the device orientation signal, to produce a first audio signal channel of a multi-channel audio signal. The first and second microphone signals are also adaptively mapped, based on the device orientation signal, to produce a second audio signal channel of a multi-channel audio signal. 1. A method of adaptively mapping a plurality of microphone signals to a multi-channel audio signal , the method comprising:obtaining at least first and second microphone signals from respective first and second spaced apart microphones;obtaining a device orientation signal from a device orientation sensor;adaptively mapping the first and second microphone signals to produce a first audio signal channel of a multi-channel audio signal, based on the device orientation signal; andadaptively mapping the first and second microphone signals to produce a second audio signal channel of a multi-channel audio signal, based on the device orientation signal.2. The method of wherein the device orientation sensor is onboard the device bearing the microphones.3. The method of claim 1 , wherein the microphone signals are obtained directly from the microphones.4. The method of claim 1 , further comprising outputting the multi-channel audio signal to a recording medium claim 1 , so as to record a multi-channel audio signal from the microphones.5. The method of claim 1 , wherein the microphone signals are obtained indirectly via an intermediate signal path claim 1 , and wherein a device orientation signal obtained contemporaneously with the microphone signals is used in the ...

METHOD AND SYSTEM FOR ENCODING AND DECODING DATA IN AUDIO

A method and system for encoding data in audio are provided. A sequence of time deltas is generated at least partially based on a set of data. At least some of the time deltas are less than a threshold at which a human naturally detects an echo. A second audio channel is generated from a first audio channel, the second audio channel being temporally shifted relative to the first audio channel using the sequence of time deltas. The first and second audio channels are played back simultaneously via at least one audio transducer. The composite audio channel is registered via at least one microphone and processed to identify the first and second audio channels that are at least partially relatively temporally shifted. A sequence of time deltas by which the second audio channel is shifted temporally relative to the first audio channel is determined, and a set of data is decoded at least partially therefrom. 1. A method for encoding data in audio , comprising:generating, via at least one processor, a sequence of time deltas at least partially based on a set of data to be encoded, at least some of the time deltas being less than a threshold at which a human naturally detects an echo;generating, from a first audio channel, a second audio channel that is at least partially temporally shifted relative to the first audio channel using the sequence of time deltas; andplaying back the first audio channel and the second audio channel simultaneously via at least one audio transducer.2. The method of claim 1 , wherein the first audio channel and the second audio channel are generated from a source audio channel.3. The method of claim 2 , wherein one of the first audio channel and the second audio channel is the source audio channel.4. The method of claim 1 , wherein claim 1 , during the generating the sequence of time deltas claim 1 , the time deltas are generated at least partially based on the frequencies of at least one of the first audio channel and the second audio channel.5. ...

Sound Localization for an Electronic Call

Headphones provide binaural sound to a user wearing the headphones. The headphones store HRTFs of the user and include head tracking and a digital signal processor (DSP). The DSP processes sound into binaural sound that externally localizes to a sound localization point (SLP) at least one meter away from a head of the user. The headphones receive head gestures from the user. 120.-. (canceled)21. A method executed by headphones to externally localize sound to a user wearing the headphones , the method comprising:storing, in a memory of the headphones, head-related transfer functions (HRTFs) of the user wearing the headphones;receiving, at a network interface of the headphones, a wireless transmission that includes the sound;tracking head movements of the user with at least one of an accelerometer, a gyroscope, and a magnetometer in the headphones;processing, with a digital signal processor (DSP) in the headphones, the sound with the HRTFs selected based on the head movements of the user in order to process the sound into binaural sound that externally localizes to the user at a sound localization point (SLP) in empty space at least one meter away from a head of the user while the head of the user moves with respect to the SLP;providing, through speakers in the headphones, the binaural sound that externally localizes to the user at the SLP in empty space at least one meter away from the head of the user while the head of the user moves with respect to the SLP;receiving, at the headphones, a head gesture from the head of the user; andaltering, by the headphones, the binaural sound in response to receiving the head gesture from the head of the user.22. The method of claim 21 , wherein the head gesture from the head of the user is received at a kinetic user interface of the headphones.23. The method of claim 21 , wherein altering the binaural sound in response to receiving the head gesture from the head of the user includes altering the binaural sound from a game that ...

Spatial audio Capture, Transmission and Reproduction

An apparatus including circuitry configured for: receiving at least two audio signals; determining at least one lower frequency effect parameter based on the at least two audio signals; determining at least one transport audio signal based on the at least two audio signals; controlling a transmission/storage of the at least one transport audio signal and the at least one lower frequency effect information such that a rendering based on the at least one transport audio signal and the at least one lower frequency effect information enables a determination of at least one low frequency effect channel. 1. An apparatus comprising:at least one processor; andat least one non-transitory memory including computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus to:receive at least two audio signals;determine at least one lower frequency effect information based on the at least two audio signals;determine at least one transport audio signal based on the at least two audio signals; andcontrol a transmission/storage of the at least one transport audio signal and the at least one lower frequency effect information such that a rendering based on the at least one transport audio signal and the at least one lower frequency effect information enables a determination of at least one low frequency effect channel.2. The apparatus as claimed in claim 1 , wherein the at least one memory and the computer program code are configured to claim 1 , with the at least one processor claim 1 , cause the apparatus to:determine at least one spatial metadata parameter based on the at least two audio signals, and wherein the apparatus is caused to control the transmission/storage of the at least one transport audio signal and the at least one lower frequency effect information is further caused to control a transmission/storage of the at least one spatial metadata parameter.3. The apparatus as claimed in claim 2 , ...

Method for Determining a Stereo Signal

A method for determining an output stereo signal comprising determining a first differential signal and determining a second differential signal; determining a first power spectrum based on the first differential signal and determining a second power spectrum based on the second differential signal; determining a first weighting function and a second weighting function as a function of the first power spectrum and the second power spectrum; and filtering a first signal, which represents a first combination of the first input audio channel signal and the second input audio channel signal, and filtering a second signal, which represents a second combination of the first input audio channel signal and the second input audio channel signal.

METHOD FOR CREATING A VIRTUAL ACOUSTIC STEREO SYSTEM WITH AN UNDISTORTED ACOUSTIC CENTER

A system and method are described for transforming stereo signals into mid and side components xm and xs to apply processing to only the side-component xs and avoid processing the mid-component. By avoiding alteration to the mid-component XM, the system and method may reduce the effects of ill-conditioning, such as coloration that may be caused by processing a problematic mid component xwhile still performing crosstalk cancellation and/or generating virtual sound sources. Additional processing may be separately applied to the mid and side components xand xs and/or particular frequency bands of the original stereo signals to further reduce ill-conditioning. 1. A method for generating a set of virtual sound sources based on a left audio signal and a right audio signal corresponding to left and right channels for a piece of sound program content , comprising:transforming the left and right audio signals to a mid-component signal and a side-component signal;generating a set of filter values for the mid-component signal and the side-component signal, wherein the filter values 1) provide crosstalk cancellation between two speakers and 2) simulate virtual sound sources for the left and right channels of the piece of sound program content;normalizing the set of filter values such that the filter values corresponding to the mid-component signal avoid altering the mid-component signal; andapplying the normalized set of filter values to one or more of the mid-component signal and the side-component signal.2. The method of claim 1 , wherein the mid-component signal is the sum of the right and left audio signals and the side-component signal is the difference between the left and right audio signals.3. The method of claim 1 , further comprising:transforming the resulting signals produced from the application of the set of normalized filter values to the one or more of the mid-component signal and the side-component signal to produce left and right filtered stereo audio signals; ...

METHOD FOR DETERMINING A PHASE FILTER FOR A SYSTEM FOR GENERATING VIBRATIONS PERCEPTIBLE BY A USER COMPRISING SEVERAL TRANSDUCERS

A determining method that makes it possible to determine a phase filter for a system that generates vibrations and that includes a first transducer for converting a first electrical signal and a second transducer for converting a second electrical signal. The method includes: performing, for at least one perception position, of a plurality of spectral measurements of a characteristic parameter of the vibrations generated in this perception position as a function of the frequency, each spectral measurement being done for a respective phase shift value between the first electrical signal and the second electrical signal; and determining a phase filter from spectral measurements done, by selecting, for each frequency, a phase shift value from among the phase shift values used to perform the spectral measurements. 1. A method for determining a phase filter for a system for generating vibrations perceptible by a user comprising a signal source configured to deliver a first electrical signal and a second electrical signal , and a pair of transducers comprising a first transducer for converting the first electrical signal into vibrations perceptible by a user and a second transducer for converting the second electrical signal into vibrations perceptible by a user , the phase filter being provided to introduce a relative phase shift between the first electrical signal and the second electrical signal , the method comprising:performing, for at least one determined perception position, of a plurality of spectral measurements of a characteristic parameter of the vibrations generated in this perception position as a function of the frequency, each spectral measurement being done for a respective phase shift value between the first electrical signal and the second electrical signal, anddetermining a phase filter from spectral measurements done, by selecting, for each frequency, a phase shift value from among the phase shift values used to perform the spectral measurements.2. The ...

Virtual Stereo Synthesis Method and Apparatus

A virtual stereo synthesis method includes acquiring at least one sound input signal on a first side and at least one sound input signal on a second side, separately performing ratio processing on a preset head related transfer function (HRTF) left-ear component and a preset HRTF right-ear component of each sound input signal on the second side, to obtain a filtering function of each sound input signal on the second side, separately performing convolution filtering on each sound input signal on the second side and the filtering function of the sound input signal on the second side, to obtain the filtered signal on the second side, and synthesizing all of the sound input signals on the first side and all of the filtered signals on the second side into a virtual stereo signal where the method may alleviate a coloration effect, and reduce calculation complexity. 1. A virtual stereo synthesis method , comprising:acquiring at least one sound input signal on a first side and at least one sound input signal on a second side;separately performing ratio processing on a preset head related transfer function (HRTF) left-ear component and a preset HRTF right-ear component of each sound input signal on the second side, to obtain a filtering function of each sound input signal on the second side;separately performing convolution filtering on each sound input signal on the second side and the filtering function of the sound input signal on the second side, to obtain a filtered signal on the second side; andsynthesizing all of the sound input signals on the first side and all of the filtered signals on the second side into a virtual stereo signal.2. The method according to claim 1 , wherein separately performing the ratio processing on the preset HRTF left-ear component and the preset HRTF right-ear component of each sound input signal on the second side claim 1 , to obtain the filtering function of each sound input signal on the second side comprises:separately using a ratio of a ...

METHOD AND MOBILE DEVICE FOR PROCESSING AN AUDIO SIGNAL

A method for processing an audio signal includes: decomposing an audio signal comprising spatial information into a set of audio signal components; and processing a first subset of the set of audio signal components according to a first processing scheme and processing a second subset of the set of audio signal components according to a second processing scheme different from the first processing scheme, wherein the first subset comprises audio signal components corresponding to at least one frontal signal source and the second subset comprises audio signal components corresponding to at least one ambient signal source; and wherein the second processing scheme is based on crosstalk cancellation. 1. A method for processing an audio signal , the method comprising:decomposing an audio signal comprising spatial information into a set of audio signal components;processing a first subset of the set of audio signal components according to a first processing scheme and processing a second subset of the set of audio signal components according to a second processing scheme different from the first processing scheme;wherein the first subset comprises audio signal components corresponding to at least one frontal signal source and the second subset comprises audio signal components corresponding to at least one ambient signal source; andwherein the second processing scheme is based on crosstalk cancellation.2. The method of claim 1 , wherein decomposing the audio signal is based on Principal Component Analysis.3. The method of claim 1 , wherein the second processing scheme is based on Head-Related Transfer Function processing.4. The method of claim 1 , wherein the first processing scheme comprises amplitude panning.5. The method of claim 1 , wherein the first processing scheme comprises delay and gain compensation.6. The method of claim 1 , wherein the first subset and the second subset of the set of audio signal components each comprise a first part associated with a left ...

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 for processing audio on an electronic device comprising a plurality of loudspeakers , the loudspeakers arranged in more than one dimension of the electronic device , comprising: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; anddispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component,wherein the method further comprises decomposing the received audio streams to direct and diffuse parts; andin determining the orientation dependent component of the rendering component, different orientation dependent components are used for the direct and diffuse parts, respectively.2. The method according to claim 1 , further comprising upmixing or downmixing the received audio streams depending on the number of the loudspeakers.3. The method ...

MULTIMEDIA APPARATUS, AND METHOD FOR PROCESSING AUDIO SIGNAL THEREOF

Disclosed in the present specification are a multimedia apparatus for servicing an optimized sound depending on the surrounding environment and a method for processing an audio signal thereof. The method for processing an audio signal of the multimedia apparatus, according to the present invention, comprises the steps of: receiving an external test sound; analyzing the frequency properties of the received test sound; calculating a compensation value of the test sound according to the analyzed frequency properties; compensating an audio signal to be output, according to the calculated compensation value; and outputting the compensated audio signal. 1receiving an external test sound;analyzing frequency characteristic of the received test sound;calculating a compensation value of the test sound according to the analyzed frequency characteristic;correcting an audio signal to output according to the calculated compensation value; andoutputting the corrected audio signal.. A method of processing an audio signal in a multimedia device, comprising: The present invention relates to a multimedia device, and more particularly, to a multimedia device and method of processing an audio signal therein, suitable for servicing optimized sound according to an ambient environment.Generally, a multimedia device is a device equipped with a function of receiving and processing a broadcast video a user can view.And, the multimedia device can display a broadcast, which is selected by a user from broadcast signals broadcasted by a broadcasting station, on a display.The global tendency of current broadcasting is toward digital broadcasting from analog broadcasting.Herein, the digital broadcasting means broadcasting of sending digital video and audio signals. Compared to analog broadcasting, digital broadcasting has less data loss owing to its strong resistance to external noise. The digital broadcasting is more advantageous in error correction, has high resolution, and provides a clear ...

PROCESSING METHOD AND SYSTEM OF AUDIO MULTICHANNEL OUTPUT SPEAKER, AND MOBILE PHONE

A processing method and a processing system of an audio multichannel output speaker and a mobile phone are disclosed. The method includes: retrieving an audio and video file and establishing four channel signals for audio signals in the audio and video file with independent channel indications; performing frequency-division processing on the four channel signals respectively, and storing data of the four channel signals after frequency-division processing in order; performing digital-to-analog conversion on the data of the four channel signals respectively, and respectively sending converted analog signal data to speakers corresponding to the four channel signals one-to-one. 1. A mobile phone with a processing system of an audio multichannel output speaker , comprising a processor and a memory , wherein the memory is configured to store computer instructions , the processor executes the computer instructions and is configured to:retrieve an audio and video file, and respectively establish four channel signals for audio signals in the audio and video file with independent channel indications;perform a frequency-division processing on the four channel signals respectively, and store data of the four channel signals after frequency-division processing in order; andperform a digital-to-analog conversion on the data of the four channel signals respectively, and respectively send converted analog signal data to speakers corresponding to the four channel signals one-to-one.2. The mobile phone according to claim 1 , wherein the processor is further configured to:retrieve the audio and video file, and divide frequencies of multichannel audio data in the audio and video file by an encoding and decoding standard;distinguish different audio channels by different channel indications, and divide the audio channels into a left front channel, a right front channel, a left rear channel, and a right rear channel;apply and create four frame buffers in an audio flinger layer; ...

AN APPARATUS AND METHOD OF AUDIO STABILIZING

A method comprising: determining a spatial audio signal; determining an apparatus motion parameter; and stabilizing the spatial audio signal dependent on the apparatus motion parameters. 124-. (canceled)25. A method comprising:determining a spatial audio signal;determining an apparatus motion parameter; andstabilizing the spatial audio signal dependent on the apparatus motion parameters.26. The method as claimed in claim 25 , further comprising:capturing at least two video images at different time instances;determining from the at least two video images a video image motion value; andwherein determining an apparatus motion parameter comprises determining an apparatus motion parameter based on the video image motion value.27. The method as claimed in claim 26 , wherein determining from the at least two video images a video image motion value further comprises:determining at least two local motion values; andcombining the at least two local motion values to generate the video image motion value.28. The method as claimed in claim 25 , wherein determining a spatial audio signal comprises:capturing at least one audio signal; anddetermining from the at least one audio signal at least one audio signal source location relative to a capturing apparatus.29. The method as claimed in claim 28 , wherein stabilizing the spatial audio signal dependent on the apparatus motion parameters comprises changing the at least one audio signal source location relative to the capturing apparatus dependent on the video image motion value.30. The method as claimed in claim 25 , wherein determining an apparatus motion parameter comprises filtering the apparatus motion parameter to remove lower frequency values.31. An apparatus comprising at least one processor and at least one memory including computer code for one or more programs claim 25 , the at least one memory and the computer code configured to with the at least one processor cause the apparatus at least to:determine a spatial audio ...

Adjusting ambient sound playing through speakers in headphones

During an electronic call between two individuals, a sound localization point simulates a location in empty space from where an origin of a voice of one individual occurs for the other individual. 120.-. (canceled)21. A method comprising:capturing, with one or more microphones in headphones worn on a head of a user, ambient sound in an environment of the user;playing, through speakers in the headphones, the ambient sound captured with the one or more microphones and music received from a smartphone in wireless communication with the headphones;determining, by the headphones, an atmospheric pressure in the environment of the user; andadjusting an amplitude of the ambient sound playing through the speakers in the headphones in response to determining the atmospheric pressure in the environment of the user.22. The method of claim 21 , wherein the adjusting of the amplitude of the ambient sound includes automatically reducing the amplitude of the ambient sound playing through the speakers in the headphones in response to determining the atmospheric pressure in the environment of the user.23. The method of further comprising:determining a global positioning system (GPS) location of the smartphone; andadjusting the amplitude of the ambient sound playing through the speakers in the headphones based on the GPS location of the smartphone.24. The method of further comprising:determining, with one or more sensors in the smartphone, motion of the smartphone with the user; andadjusting the amplitude of the ambient sound playing through the speakers in the headphones based on the motion of the smartphone.25. The method of further comprising:determining, with an accelerometer in the smartphone, motion of the smartphone with the user; andcanceling non-vocal components of the ambient sound playing through the speakers in the headphones in response to determining the motion of the smartphone.26. The method of further comprising:determining, from the ambient sound, a voice in the ...

Apparatus for managing distortion in a signal path and method

The transparent conductive film includes on a transparent film base a transparent electrode layer in which a transparent conductive oxide layer and a patterned metal layer are stacked in contact with each other. The maximum layer thickness of the transparent electrode layer is 300 nm or less. The metal layer has a metal pattern width of 1 μm or more and 8 μm or less, and the metal pattern coverage ratio of 0.4% or more and 3.2% or less. It is preferable that the metal layer has a layer thickness of 50 nm or more and 250 nm or less. It is also preferable that the pattern shape of the metal layer is of stripes, mesh, dots or the like.

AUDIO PROCESSOR FOR ORIENTATION-DEPENDENT PROCESSING

Audio processor having an input interface, a detector interface, a mixer and an output interface. The input interface receiving at least two input audio channels, each input audio channel being associated with a predetermined reproduction position of at least two loudspeakers on at least one loudspeaker axis. The detector interface receiving a position signal indicating an information on a position of the at least two loudspeakers with respect to an ear axis of a listener, wherein the ear axis and the at least one loudspeaker axis have an angle to each other, being greater than 0° and lower than 180°. The mixer mixing the at least two input audio channels to obtain the at least two output channels depending on the position signal. 1. Audio processor comprising:an input interface for receiving at least two input audio channels, each input audio channel being associated with a predetermined reproduction position of at least two loudspeakers on at least one loudspeaker axis;a detector interface for receiving a position signal indicating an information on a position of the at least two loudspeakers with respect to an ear axis of a listener, wherein the ear axis and the at least one loudspeaker axis have an angle to each other, being greater than 0° and lower than 180°;a mixer for mixing the at least two input audio channels to obtain the at least two output channels depending on the position signal, such thata portion of the second input audio channel in the first output channel for a first angle between the ear axis and the loudspeaker axis is greater than a portion of the second input audio channel in the first output channel for a second angle between the ear axis and the loudspeaker axis, wherein the first angle is greater than the second angle ora portion of the first input audio channel in the second output channel for the first angle is greater than a portion of the first input audio channel in the second output channel for the second angle, wherein the first ...

DEVICE AND METHOD FOR PLAYING SOUND

The present invention extracts azimuth information on a sound source, read the touch state of a touch screen on which an image is displayed, and enables a sound source having azimuth information corresponding to a place touched on the image to be synthesized so as to be distinguished from other sound sources. According to the present invention, since it is possible to listen to the distinguished sound of a desired location on an image, a user may be provided with more satisfaction. 1. A sound reproducing apparatus comprising:an azimuth information extractor to extract azimuth information of a sound source from at least one stereo sound information;a touch screen to display an image;an image controller to read out at least a touch state of the touch screen; anda synthesizer to characterize and synthesize a sound source, which has azimuth information corresponding to a touch point on the image, according to the touch state by differentiating the sound source from other sound sources.2. The sound reproducing apparatus of claim 1 , further comprising an azimuth calculator to calculate touch information of the touch point on the image claim 1 , which includes at least the azimuth information corresponding to the touch point based on the touch state read out by the image controller.3. The sound reproducing apparatus of claim 2 , wherein the touch information comprises area information.4. The sound reproducing apparatus of claim 2 , wherein azimuth information included in information of the image and the azimuth information of the sound source are scaled such that the azimuth information included in the information of the image is matched with the azimuth information of the sound source.5. The sound reproducing apparatus of claim 1 , further comprising a memory to store the at least one stereo sound information and at least one image.6. The sound reproducing apparatus of claim 1 , wherein the at least one stereo sound information and at least one image are transmitted ...

LOUDNESS ADJUSTMENT FOR DOWNMIXED AUDIO CONTENT

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content. 1. A method for gain adjusting audio signals based on encoder-generated loudness metadata , the method comprising:receiving, by an audio decoder operating in a playback channel configuration different from a reference channel configuration, an audio signal for the reference channel configuration, the audio signal including audio sample data for each channel of the reference channel configuration, and the encoder-generated loudness metadata, the encoder-generated loudness metadata comprising loudness metadata for a plurality of channel configurations including the playback channel configuration and the reference channel configuration;downmixing the audio sample data into downmixed audio sample data for the audio channels of the playback channel configuration;selecting, from the loudness metadata for the plurality of channel configurations, the loudness metadata for the playback channel configuration;determining loudness adjustment gains from the loudness metadata for the playback channel configuration; andapplying the loudness adjustment gains as a part of overall gains applied to the downmixed audio sample data to generate output audio sample data for each channel of the playback channel configuration;wherein the loudness adjustment gains depend on a loudness level indicated by the loudness metadata for the ...

Modified Directional Effect

An example method is performed by a media playback system comprising a plurality of audio drivers configured to output audio content according to a first radiation pattern that produces an inherent directional effect. Based on data indicating characteristics of a listening area in which the media playback system operates, the system determines first and second transfer functions corresponding to the first and second audio drivers, respectively. One or both of the transfer functions configure the first and second audio drivers to output audio content according to a second radiation pattern that produces a modified directional effect relative to the first radiation pattern. The system applies the transfer function to audio content thereby causing the first and second audio drivers to play back audio content according to the second radiation pattern. 1. A media playback system comprising:a processor;a network interface;a first audio driver and a second audio driver, wherein the first and second audio drivers are configured to output audio content according to a first radiation pattern that produces an inherent directional effect; and receiving, via the network interface, first data and second data representing first and second channels, respectively, of multi-channel audio content, wherein the first and second data comprises audio data over a range of frequencies;', 'receiving third data indicating one or more characteristics of a listening area in which the media playback system operates;', 'determining, based on the third data, a first transfer function corresponding to the first audio driver and a second transfer function corresponding to the second audio driver, wherein at least one of the first and second transfer functions configure the first and second audio drivers to output audio content according to a second radiation pattern that produces a modified directional effect relative to the first radiation pattern;', 'generating a first audio output signal and a ...

Methods and apparatus for processing stereophonic audio content

A method of processing stereophonic audio content received in a first audio channel to be output to a first speaker and a second audio channel to be output to a second speaker, the method comprising: receiving the first and second audio channels; identifying a plurality of frequency sub-bands in the first audio channel; for each of the plurality of frequency sub-bands, determining an importance weighting based on a degree of audibility of the sub-band when combined with the remainder of sub-bands of the first audio channel and the second audio channel; on determining that a peak amplitude of the first audio channel is above a first clipping threshold, iteratively suppressing the sub-band of least importance in the first audio signal until the peak amplitude of the first audio signal is below the first clipping threshold; and outputting the suppressed first audio channel.

SOUND OUTPUT CIRCUIT, PROJECTOR, AND CONTROL METHOD OF SOUND OUTPUT CIRCUIT

A sound output circuit includes an amplifier that amplifies a sound signal, a first terminal for external connection connected to an output terminal of the amplifier, a reference voltage output that outputs a reference voltage, a second terminal for external connection, and a controller that controls a conduction state of an output terminal of the reference voltage output and the second terminal for external connection according to a voltage of the second terminal for external connection. 1. A sound output circuit comprising:an amplifier that amplifies a sound signal;a first terminal for external connection connected to an output terminal of the amplifier;a reference voltage output that outputs a reference voltage;a second terminal for external connection; anda controller that controls a conduction state of an output terminal of the reference voltage output and the second terminal for external connection according to a voltage of the second terminal for external connection.2. The sound output circuit according to claim 1 , wherein the controller sets the output terminal of the reference voltage output and the second terminal for external connection in a non-conduction state when the voltage of the second terminal for external connection is a ground voltage.3. The sound output circuit according to claim 1 , wherein the controller includes:a switch connected between the output terminal of the reference voltage output and the second terminal for external connection; anda switch controller that controls a conduction state of the switch according to the voltage of the second terminal for external connection.4. The sound output circuit according to claim 3 , wherein the switch is an FET.5. The sound output circuit according to claim 3 , further comprising an integrated circuit on which the first terminal for external connection claim 3 , the second terminal for external connection claim 3 , and the controller are not mounted claim 3 , but the amplifier and the reference ...