УСОВЕРШЕНСТВОВАННОЕ СТЕРЕОФОНИЧЕСКОЕ КОДИРОВАНИЕ НА ОСНОВЕ КОМБИНАЦИИ АДАПТИВНО ВЫБИРАЕМОГО ЛЕВОГО/ПРАВОГО ИЛИ СРЕДНЕГО/ПОБОЧНОГО СТЕРЕОФОНИЧЕСКОГО КОДИРОВАНИЯ И ПАРАМЕТРИЧЕСКОГО СТЕРЕОФОНИЧЕСКОГО КОДИРОВАНИЯ

Изобретение относится к средствам для кодирования и декодирования звуковых сигналов. Технический результат заключается в повышении эффективности кодирования. Генерируют низведенный сигнал и остаточный сигнал на основе стереофонического сигнала. Генерируют SBR-кодированный низведенный сигнал и монофонические параметры SBR в ответ на низведенный сигнал. Определяют один или несколько параметров параметрического стереофонического кодирования. Перцептуально кодируют в нисходящем направлении относительно генерирования SBR-кодированного низведенного сигнала и остаточного сигнала. При этом кодирование на основе суммы SBR-кодированного низведенного сигнала и остаточного сигнала и на основе разности SBR-кодированного низведенного сигнала и остаточного сигнала выбирают зависящим от частоты или независящим от частоты образом. При этом указанный способ позволяет выбирать между параметрическим стереофоническим кодированием или левым/правым кодированием стереофонического сигнала в сигнал битового потока ...

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

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

УСОВЕРШЕНСТВОВАННОЕ СТЕРЕОФОНИЧЕСКОЕ КОДИРОВАНИЕ НА ОСНОВЕ КОМБИНАЦИИ АДАПТИВНО ВЫБИРАЕМОГО ЛЕВОГО/ПРАВОГО ИЛИ СРЕДНЕГО/ПОБОЧНОГО СТЕРЕОФОНИЧЕСКОГО КОДИРОВАНИЯ И ПАРАМЕТРИЧЕСКОГО СТЕРЕОФОНИЧЕСКОГО КОДИРОВАНИЯ

Изобретение относится к средствам кодирования и декодирования звуковых сигналов. Технический результат заключается в повышении качества кодирования сигнала. Генерируют низведенный сигнал и остаточный сигнал на основе стереофонического сигнала. Определяют разность интенсивностей между каналами и взаимную корреляцию между каналами. Предпочтительно, параметры параметрического стереофонического кодирования являются зависящими от времени и от частоты. Этап преобразования генерирует псевдолевый/правый стереофонический сигнал путем выполнения преобразования на основе низведенного сигнала и остаточного сигнала. Псевдостереофонический сигнал обрабатывается перцептуальным стереофоническим кодером. Для стереофонического кодирования может быть выбрано левое/правое кодирование или среднее/побочное кодирование. Предпочтительно, выбор между левым/правым кодированием и средним/побочным кодированием является зависящим от времени и от частоты. 8 н.з. и 58 з.п. ф-лы, 26 ил.

ДЕКОРРЕЛЯЦИЯ СИГНАЛОВ В СИСТЕМЕ ОБРАБОТКИ АУДИОДАННЫХ

Изобретение относится к области обработки сигнала. Техническим результатом является уменьшение сложности алгоритмов кодирования и декодирования за счет декорреляции только вещественнозначных коэффициентов. Способ обработки аудиоданных включает прием из битового потока аудиоданных, соответствующих ряду звуковых каналов, применение процесса декорреляции к некоторым из аудиоданных, причем процесс декорреляции включает применение алгоритма декорреляции, действующего полностью на вещественнозначных коэффициентах. 3 н. и 13 з.п. ф-лы, 37 ил.

ПОСЕГМЕНТНАЯ НАСТРОЙКА ПРОСТРАНСТВЕННОГО АУДИОСИГНАЛА К ДРУГОЙ УСТАНОВКЕ ГРОМКОГОВОРИТЕЛЯ ДЛЯ ВОСПРОИЗВЕДЕНИЯ

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

СПОСОБЫ УПРАВЛЕНИЯ МЕЖКАНАЛЬНОЙ КОГЕРЕНТНОСТЬЮ ЗВУКОВЫХ СИГНАЛОВ, ПОДВЕРГНУТЫХ ПОВЫШАЮЩЕМУ МИКШИРОВАНИЮ

Изобретение относится к средствам для управления межканальной когерентностью звуковых сигналов, подвергнутых повышающему микшированию. Технический результат заключается в повышении качества кодирования и декодирования аудиоданных. Определяют характеристики аудиоданных, соответствующих ряду звуковых каналов. Эти звуковые характеристики могут содержать данные пространственных параметров. Процессы декорреляционной фильтрации для аудиоданных могут основываться на звуковых характеристиках. Эти процессы декорреляционной фильтрации могут вызывать специфичную когерентность между сигналами декорреляции между специфичными для каналов сигналами декорреляции для по меньшей мере одной пары каналов. Такие специфичные для каналов сигналы декорреляции могут приниматься и/или определяться. Можно управлять межканальной когерентностью между рядом пар звуковых каналов. Путем выполнения операций на фильтрованных аудиоданных можно синтезировать набор специфичных для каналов сигналов декорреляции, соответствующих ...

СПОСОБ, УСТРОЙСТВО, КОДИРУЮЩЕЕ УСТРОЙСТВО, ДЕКОДИРУЮЩЕЕ УСТРОЙСТВО И АУДИОСИСТЕМА

Изобретение относится к обработке стереосигнала, полученного от кодировщика. Способ обработки стереосигнала включает в себя этапы, на которых кодируют N-канальный аудиосигнал в стереосигнал (L0, R0) и пространственные параметры (w1, wr), обрабатывают стереосигнал с использованием пространственных параметров для генерации обработанного стереосигнала (L0w, R0w). Матрица обработанного стереосигнала может быть описана как матрица стереосигнала, умноженная на фильтр-матрицу (Н), элементы которой являются фильтр-функциями (Н1, Н2, Н3, Н4), работающими с пространственными параметрами (w1, wr) и константой (а). Фильтр-функции не зависят от времени и выбираются так, чтобы матрица была обратимой. Технический результат - улучшение понижающего микширования в плане качества восприятия или его пространственных свойств. 7 н. и 13 з.п. ф-лы, 4 ил.

УСТРОЙСТВО ПРЕДОСТАВЛЕНИЯ АУДИО И СПОСОБ ПРЕДОСТАВЛЕНИЯ АУДИО

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

УЛУЧШЕНИЕ ЗВУКОВОГО СИГНАЛА FM-СТЕРЕОФОНИЧЕСКОГО РАДИОПРИЕМНИКА ПУТЕМ ИСПОЛЬЗОВАНИЯ ПАРАМЕТРИЧЕСКОГО СТЕРЕОФОНИЧЕСКОГО КОДИРОВАНИЯ

Изобретение относится к устройству для улучшения стереофонического звукового сигнала FM-стереофонического радиоприемника. Устройство включает этап оценки параметров параметрического стереофонического кодирования (PS). Этап оценки параметров сконфигурирован для определения одного или нескольких параметров параметрического стереофонического кодирования на основе стереофонического звукового сигнала зависящим от частоты или не зависящим от частоты образом. Указанные параметры PS предпочтительно являются зависящими от времени и от частоты. Кроме того, устройство включает этап повышающего микширования. Этап повышающего микширования сконфигурирован для генерирования улучшенного стереофонического сигнала на основе первого звукового сигнала и одного или нескольких параметров параметрического стереофонического кодирования. Первый звуковой сигнал получается из стереофонического звукового сигнала, например, посредством операции понижающего микширования на этапе понижающего микширования. Этап оценки ...

УСТРОЙСТВО И СПОСОБ ГЕНЕРИРОВАНИЯ СТЕРЕОСИГНАЛА

... 1. Устройство генерирования стереосигнала, содержащее: блок преобразования, который преобразует монофонический сигнал во временной области, полученный из сигналов правого и левого каналов стереосигнала, в монофонический сигнал в частотной области; блок вычисления энергии, который определяет первый энергетический спектр монофонического сигнала в частотной области; блок вычисления коэффициента масштабирования, который определяет первый коэффициент масштабирования для энергетического спектра левого канала стереосигнала из первой разности между первым энергетическим спектром и энергетическим спектром левого канала стереосигнала и определяет второй коэффициент масштабирования для правого канала из второй разности между первым энергетическим спектром и энергетическим спектром правого канала стереосигнала; и блок умножения, который умножает монофонический сигнал в частотной области на первый коэффициент масштабирования для генерирования сигнала левого канала стереосигнала и умножает монофонический ...

СПОСОБ, УСТРОЙСТВО, КОДИРУЮЩЕЕ УСТРОЙСТВО, ДЕКОДИРУЮЩЕЕ УСТРОЙСТВО И АУДИОСИСТЕМА

... 1. Способ обработки стереосигнала, полученного от кодировщика, где кодировщик кодирует N-канальный аудиосигнал в сигналы левого и правого каналов (L0;R0) и пространственные параметры (P), способ включает в себя этап, на котором обрабатывают упомянутые сигналы левого и правого каналов для того, чтобы обеспечить обработанные сигналы (L0w;R0w), в которых упомянутая обработка управляется в зависимости от упомянутых пространственных параметров (P). 2. Способ по п.1, в котором упомянутая обработка управляется первым параметром (wl;wr) для каждого из упомянутых сигналов левого и правого каналов, упомянутый первый параметр зависит от пространственных параметров (P). 3. Способ по п.2, в котором упомянутый первый параметр (wl;wr) является функцией времени и/или частоты. 4. Способ по п.1, в котором упомянутая обработка включает в себя фильтрацию, по меньшей мере, одного из упомянутых сигналов левого и правого каналов функцией преобразования, которая зависит от пространственных параметров (P). 5. Способ ...

УСТРОЙСТВО ОБРАБОТКИ СИГНАЛОВ ДЛЯ УСИЛЕНИЯ РЕЧЕВОГО КОМПОНЕНТА В МНОГОКАНАЛЬНОМ ЗВУКОВОМ СИГНАЛЕ

Изобретение относится к средствам для обработки сигналов для усиления речевого компонента в многоканальном звуковом сигнале. Технический результат заключается в повышении эффективности усиления речевого компонента в многоканальном звуковом сигнале. Определяют меру, представляющую общую величину многоканального звукового сигнала по частоте, на основе звукового сигнала левого канала, звукового сигнала центрального канала, и звукового сигнала правого канала. Получают функцию коэффициента усиления на основе соотношения между мерой величины звукового сигнала центрального канала и мерой, представляющей общую величину многоканального звукового сигнала. Взвешивают звуковой сигнал левого канала посредством функции коэффициента усиления для получения взвешенного звукового сигнала левого канала. Взвешивают звуковой сигнал центрального канала посредством функции коэффициента усиления для получения взвешенного звукового сигнала центрального канала. Взвешивают звуковой сигнала правого канала посредством ...

УСТРОЙСТВО ПРЕДОСТАВЛЕНИЯ АУДИО И СПОСОБ ПРЕДОСТАВЛЕНИЯ АУДИО

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

УСОВЕРШЕНСТВОВАННОЕ СТЕРЕОФОНИЧЕСКОЕ КОДИРОВАНИЕ НА ОСНОВЕ КОМБИНАЦИИ АДАПТИВНО ВЫБИРАЕМОГО ЛЕВОГО/ПРАВОГО ИЛИ СРЕДНЕГО/ПОБОЧНОГО СТЕРЕОФОНИЧЕСКОГО КОДИРОВАНИЯ И ПАРАМЕТРИЧЕСКОГО СТЕРЕОФОНИЧЕСКОГО КОДИРОВАНИЯ

УСТРОЙСТВО И СПОСОБ ДЛЯ ФОРМИРОВАНИЯ ЗАКОДИРОВАННОГО СТЕРЕОСИГНАЛА АУДИОЧАСТИ ИЛИ ПОТОКА ДАННЫХ АУДИО

... 1. Устройство для формирования кодированного стереосигнала аудиочасти или потока данных аудио, имеющего первый стереоканал и второй стереоканал из многоканального представления аудио части или потока данных аудио, содержащего информацию относительно более двух множественных каналов, содержащее: ! средство (11) для обеспечения более двух множественных каналов из многоканального представления; ! средство (12) для выполнения обработки сигналов головного телефона для формирования некодированного стереосигнала с некодированным первым стереоканалом (10a) и некодированным вторым стереоканалом (10b), причем средство (12) для выполнения сформировано для того, чтобы ! оценивать каждый множественный канал посредством первой функции фильтра (HiL), полученной из виртуальной позиции громкоговорителя для воспроизведения этого множественного канала и виртуальной позиции первого уха слушателя, для первого канала стерео и второй функции фильтра (HiR), полученной из виртуальной позиции громкоговорителя и ...

ОБЪЕДИНЕНИЕ ПРОСТРАНСТВЕННОГО ЗВУЧАНИЯ ДЛЯ ОБЛАСТИ НЕСКОЛЬКИХ ИСТОЧНИКОВ ПРОСТРАНСВЕННОГО ЗВУЧАНИЯ

ПОСЕГМЕНТНАЯ НАСТРОЙКА ПРОСТРАНСТВЕННОГО АУДИОСИГНАЛА К ДРУГОЙ УСТАНОВКЕ ГРОМКОГОВОРИТЕЛЯ ДЛЯ ВОСПРОИЗВЕДЕНИЯ

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

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

VORRICHTUNG UND VERFAHREN ZUR ERZEUGUNG EINES PEGELPARAMETERS UND VORRICHTUNG UND VERFAHREN ZUR ERZEUGUNG EINER MEHRKANAL-REPRÄSENTATION

Spatial audio generator

Sound imaging system for a video game

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

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

Audio enhancement module for portable media player

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

Producing simulated sound distance effects

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

REVERBERATION APPARATUS

ANORDNUNG ZUR DRAHTLOSEN UBERTRAGUNG VON NIEDERFREQUENZSIGNALEN

PROCEDURE FOR THE AUTOMATIC ENTZERREN OF AN ACOUSTIC IRRADIATION SYSTEM

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

AUDIO CODING DEVICE AND AUDIO CODING METHOD

PRODUCTION OF DEKORRELIERTER SIGNALS

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

DEVICE AND PROCEDURE FOR PRODUCING A CODED STEREO SIGNAL OF AN AUDIO PIECE OR AUDIO DATA STREAM

TO CODING OF SOUND SIGNALS WITH RESOUND

SOUND SIGNAL PROCESSING

OHRAUFLIEGENDER KOPFHÖRER

DIALOGUE IMPROVEMENT TECHNIQUES

WITH PHASE ALLOCATION OF WORKING, STEREOPHONER SCHALLSYNTHESIZER.

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

SOUND IMAGING METHOD AND APPARATUS

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

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

REVERBERATION APPARATUS

Sound image manipulation apparatus and method for sound image enhancement

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

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

Method and 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.

Method and apparatus for providing customised sound distributions

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

Signaling layers for scalable coding of higher order ambisonic audio data

In general, techniques are described for signaling layers for scalable coding of higher order ambisonic audio data. A device comprising a memory and a processor may be configured to perform the techniques. The memory may be configured to store the bitstream. The processor may be configured to obtain, from the bitstream, an indication of a number of layers specified in the bitstream, and obtain the layers of the bitstream based on the indication of the number of layers.

SURROUND-SOUND SYSTEM

STEREOPHONIC VOICE SIGNAL TRANSMISSION SYSTEM

CENTER MODE CONTROL CIRCUIT FOR SOUND SYSTEM

A center mode control circuit for a sound system comprises an amplifying portion (31) for amplifying a center input signal, a switch (34) connected to an input terminal of the amplifying portion, capacitors (35, 36) selected by the switch and having different characteristic from each other. The switched are a normal mode for adding low frequency component of the center input signal to left and right input signals, a phantom mode for adding full range of the center input signal to the left and the right stereo input signals, and a wide mode for not adding the center input signal to the left and the right stereo input signals.

SYSTEM AND TOOLS FOR ENHANCED 3D AUDIO AUTHORING AND RENDERING

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

AUDIO SIGNAL PROCESSING SYSTEM

RECONSTRUCTING AUDIO SIGNALS WITH MULTIPLE DECORRELATION TECHNIQUES

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

MULTICHANNEL AUDIO CODING

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

ADVANCED STEREO CODING BASED ON A COMBINATION OF ADAPTIVELY SELECTABLE LEFT/RIGHT OR MID/SIDE STEREO CODING AND OF PARAMETRIC STEREO CODING

The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. More-over, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

A SIGNAL PROCESSING APPARATUS FOR ENHANCING A VOICE COMPONENT WITHIN A MULTI-CHANNEL AUDIO SIGNAL

The invention relates to a signal processing apparatus (100) for enhancing a voice component within a multi-channel audio signal, the multi-channel audio signal comprising a left channel audio signal (L), a center channel audio signal (C), and a right channel audio signal (R), the signal processing apparatus (100) comprising a filter (101) and a combiner (103); wherein the filter (101) is configured to determine a measure representing an overall magnitude of the multi-channel audio signal over frequency upon the basis of the left channel audio signal (L), the center channel audio signal (C), and the right channel audio signal (R), to obtain a gain function (G) based on a ratio between a measure of magnitude of the center channel audio signal (C) and the measure representing the overall magnitude of the multi- channel audio signal, and to weight the left channel audio signal (L) by the gain function (G) to obtain a weighted left channel audio signal (LE), to weight the center channel audio ...

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

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

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

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

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.

ENHANCED CODING AND PARAMETER REPRESENTATION OF MULTICHANNEL DOWNMIXED OBJECT CODING

An audio object coder for generating an encoded object signal using a plurality of audio objects includes a downmix information generator for generating downmix information indicating a distribution of the plurality of audio objects into at least two downmix channels, an audio object parameter generator for generating object parameters for the audio objects, and an output interface for generating the imported audio output signal using the downmix information and the object parameters. An audio synthesizer uses the downmix information for generating output data usable for creating a plurality of output channels of the predefined audio output configuration.

SOUND IMAGING APPARATUS FOR A VIDEO GAME SYSTEM

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

Vorrichtung zur Übertragung von elektrischen Schwingungen akustischer Frequenz.

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

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

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

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

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

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

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

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

CODING OF AUDIO SCENES

Spatial upmix controlled by renderer

The content reproducing multi-channel electronic device and method

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

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

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

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

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

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

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

descorrelação de sinal em um sistema de processamento de áudio

Apparatus and method for encoding/decoding signal

An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a three-dimensional (3D) down-mix signal from an input bitstream, generating a down-mix signal with 3D effects removed therefrom by performing a 3D rendering operation on the extracted 3D down-mix signal, and generating a 3D down-mix signal with 3D effects by performing a 3D rendering operation on the generated down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Removing time delays in signal paths

The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

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

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

Pseudo-stereophonic device

In this pseudo-stereophonic device to simulate the pseudo-stereophonic signal using monotonic signal, comprises m delaying units interconnected in series and inputted with delaying signal S one-by-one; m Finite Impulse Response (FIR) digital filters for performing individually filtering process of the output signals Sk (where k equals 1, 2, up to m) from each delaying unit. The output from each FIR digital filter defined as Yk. (where k equals 1, 2, up to m). The simulated pseudo-stereophonic signal Lout and Rout could be produced by the following formulas calculation: Lout=Y1+Sum of Yn (where n equals 2, 3, up to m) and Rout=Y1-Sum of Yn (where n equals 2, 3, up to m).

MAPPING VIRTUAL SPEAKERS TO PHYSICAL SPEAKERS

Techniques are described for mapping virtual speakers to physical speakers, having first adjusted the position of one of the virtual speakers based on a relative position of the one of the virtual speakers to one of the physical speakers. A device comprising one or more processors may perform the techniques. The one or more processors may be configured to determine a difference in position between one of a plurality of physical speakers and one of a plurality of virtual speakers arranged in a geometry, and adjust a position of the one of the plurality of virtual speakers within the geometry based on the determined difference in position and prior to mapping the plurality of virtual speakers to the plurality of physical speakers.

APPARATUS AND METHOD FOR SYNTHESIZING PSEUDO-STEREOPHONIC OUTPUTS FROM A MONOPHONIC INPUT

L'invention porte sur un synthétiseur euphonique permettant d'obtenir des signaux pseudo-stéréophoniques gauche (L) et droits (R) à partir d'un signal monophonique (220). Ce dernier (220) passe par un filtre (504) perspectif produisant un signal en mode différentiel, puis par un filtre égaliseur (506) qui produit un signal en mode commun. Le filtre (504) perspectif atténue les composantes du signal dans le domaine de fréquences correspondant à la voix humaine, tandis que le filtre égaliseur (506) les atténue dans le domaine de fréquences n'y correspondant pas. Le filtre égaliseur (506) produit également un déphasage de 90°. Les signaux en mode différentiel et en mode commun sont combinés pour donner les voies droite et gauche (L, R). Lesdits signaux pseudo-stéréophoniques (L, R) ainsi créés ont relativement moins d'ambiance dans le domaine de fréquences correspondant à la voix humaine, et relativement plus dans le domaine de fréquences n'y correspondant pas.

GENERATION OF A SOUND SIGNAL

The present invention relates to a method and a media system of/for generation of at least one output signal (HPL, HPR) from at least one input signal from a second set of sound signals (M) having a related second set of Head Related Transfer Functions. The media system can be a TV, a CD player, a DVD player, a Radio, a display, an amplifier, a headphone or a VCR. Said method includes the steps of determining, for each signal in the second set of sound signals, a weighted relation (14) comprising at least one signal from a third set of intermediate sound signals (CHI1, CHI2) and at least one weight value (Weights); determining a first set of Head Related Transfer Functions (HRTFs) based on the second set of sound signals, the second set of Head Related Transfer Functions and the weighted relation; and transferring at least one signal from the third set of intermediate sound signals by means of at least one HRTF from said first set of Head Related Transfer Functions in order to generate ...

CENTER CHANNEL ENHANCEMENT OF VIRTUAL SOUND IMAGES

The present invention disclosed and claimed herein, in one aspect thereof, comprises a method for enhancing the front sound image during reproduction in a listening space of a stereo sound program (Figs. 12-13), comprising the steps of receiving left and right channels (58, 60) of the stereo sound program; generating a virtual center channel signal (212) from the left and right channels (58, 60) of the stereo sound program; and driving a center channel speaker (212) with the virtual center channel signal, the center channel speaker (212) disposed at a center location in a front portion of the listening space.

Spatial-based audio object generation using image information

Methods and systems for generating a multichannel audio object. One or more features in a given video frame are identified using one or more image analysis neural networks. A multichannel audio object is generated based on the one or more identified features and one or more baseline audio tracks using an audio neural network.

Efficient and scalable parametric stereo coding for low bitrate audio coding applications

The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Multi-channel speaker utilizing dual-voice coils

A speaker that utilizes a dual voice coil speaker element to generate sound corresponding to more than one sound channel is disclosed. The speaker may be paired with a substantially identical speaker to generate stereo sound corresponding to left and right channel signals and monaural sound corresponding to a center channel signal. Such a configuration enables the speakers to generate sound corresponding to side and center channel signals without requiring the use of discrete speaker elements for each channel.

Generation of a sound signal

In a method and a media system of/for generation of at least one output signal (HPL,HPR) from at least one input signal belonging to a second set of sound signals (M) having a related second set of Head Related Transfer Functions, in which the media system can be a TV, a CD player, a DVD player, a Radio, a display, an amplifier, a headphone or a VCR, the method includes the steps of determining, for each signal in the second set of sound signals, a weighted relation (14) including at least one signal belonging to a third set of intermediate sound signals (CHI1, CHI2) and at least one weight value (Weights); determining a first set of Head Related Transfer Functions (HRTFs) based on the second set of sound signals, the second set of Head Related Transfer Functions and the weighted relation; and transferring at least one signal belonging to the third set of intermediate sound signals by means of at least one HRTF belonging to said first set of Head Related Transfer Functions in order to generate ...

SPATIAL DELIVERY OF MULTI-SOURCE AUDIO CONTENT

A system for enabling spatial delivery of multi-source audio data to a user based on a multi-layer audio stack is provided. The multi-layer audio stack includes a central layer located within a predetermined vertical distance from a reference line associated with the user, such as the horizon line of the user. The multi-layer audio stack can also include an upper layer located above the central layer and/or a lower layer located below the central layer. Audio data from multiple sources are collected and prioritized based on context data gathered for the user. Audio data on which the user would like to focus is assigned the highest priority and delivered on the central layer. Audio data that the user does not currently focus on, hut would like to visit next, can be assigned a lower priority and be delivered in the upper layer or the lower layer. The user can shift the multi-layer audio stack up or down to navigate through the audio data rendered at different layers of the stack.

Removing time delays in signal paths

The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

System and method for using endpoints to provide sound monitoring

A method is provided in one example embodiment that includes monitoring a sound pressure level with an endpoint (e.g., an Internet Protocol (IP) phone), which is configured for communications involving end users; analyzing the sound pressure level to detect a sound anomaly; and communicating the sound anomaly to a sound classification module. The endpoint can be configured to operate in a low-power mode during the monitoring of the sound pressure level. In certain instances, the sound classification module is hosted by the endpoint. In other implementations, the sound classification module is hosted in a cloud network.

METHOD AND APPARATUS FOR DOWNMIXING MULTI-CHANNEL AUDIO SIGNALS

Downmixing multi-channel audio signals to target channels by pre-downmixing frequency coefficients that are encoded using a most frequently used block type in stereo channels in the frequency domain, thereby reducing an amount of calculations and an amount of power required to downmix the multi-channel audio signals.

Simulating Acoustic Output at a Location Corresponding to Source Position Data

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

CONTENT-AWARE AUDIO MODES

An audio system for manipulating audio material for playback based on the content of the material includes an input for receiving input audio signals, an output for providing output audio signals to loudspeakers, and an audio processor that processes the input audio signals to produce the output audio signals. The audio processor determines a type of content represented by the input audio signals, selects a set of processing instructions to control the spatial presentation of the audio material based on the determined type, and applies the selected set of processing instructions to the input audio signals.

Signal Processing for a Headrest-Based Audio System

An automobile audio system having at least two near-field speakers located close to an intended position of a listener's head is configured by determining a first binaural filter that causes sound produced by each of the near-field speakers to have characteristics at the intended position of the listener's head of sound produced by a sound source located at a first designated position other than the actual locations of the near-field speakers, determining an up-mixing rule to generate at least three component channel signals from an input audio signal having at least two channels, and configuring the audio system to, determine a first binaural signal corresponding to a combination of the component channel signals originating at the first designated position, and filter the first binaural signal using the first binaural filter and to output the filtered signals using the near-field speakers.

Spatial transformation of ambisonic audio data

A device configured to decode a bitstream, where the device includes a memory configured to store a temporally encoded representation of spatial audio signals. The device is also configured to receive the bitstream that includes an indication of a spatial transformation, and includes a temporal decoding unit, coupled to the memory, configured to decode one or more spatial audio signals represented in a spatial domain, where the one or more spatial audio signals are associated with different angles in the spatial domain. In addition, the device includes an inverse spatial transformation unit, coupled to the temporal decoding unit, is configured to convert the one or more spatial audio signals represented in the spatial domain into at least three ambisonic coefficients that, in part, represent a soundfield in an ambisonics domain, and perform a spatial transformation of the soundfield based on the indication of the spatial transformation received in the bitstream.

AUDIO PROCESSING METHOD AND AUDIO PROCESSING APPARATUS

System for extraction of reverberant content of an audio signal

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

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

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

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

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

Method for Driving Loudspeakers

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

Audio signal of an fm stereo radio receiver by using parametric stereo

The invention relates to an apparatus for improving a stereo audio signal of an FM stereo radio receiver. The apparatus comprises a parametric stereo (PS) parameter estimation stage. The parameter estimation stage is configured to determine one or more parametric stereo parameters based on the stereo audio signal in a frequency-variant or frequency-invariant manner. Preferably, these PS parameters are time- and frequency-variant. Moreover, the apparatus comprises an upmix stage. The upmix stage is configured to generate the improved stereo signal based on a first audio signal and the one or more parametric stereo parameters. The first audio signal is obtained from the stereo audio signal, e.g. by a downmix operation in a downmix stage. The PS parameter estimation stage may be part of a PS encoder. The upmix stage may be part of a PS decoder.

Speaker Apparatus

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

Camera Driven Audio Spatialization

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

APPARATUS AND METHOD FOR GENERATING PANORAMIC SOUND

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

Downmix Limiting

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

AUDIO SIGNAL PROCESSING DEVICE

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

AUTOMATIC ANNOUNCER VOICE ATTENUATION IN A PRESENTATION OF A BROADCAST EVENT

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

Diffusing Acoustical Crosstalk

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

AUDIO SIGNAL REPRODUCTION DEVICE AND AUDIO SIGNAL REPRODUCTION METHOD

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

SPATIALIZATION USING STEREO DECORRELATION

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

SYSTEM AND METHOD FOR REPRODUCING WAVE FIELD USING SOUND BAR

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

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

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

NAVIGATION WITH THREE DIMENSIONAL AUDIO EFFECTS

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

USER AUTHENTICATION METHOD AND USER AUTHENTICATION DEVICE

Provided is a user authentication method including reproducing sound data of which a sound source in a first position of a space around a user is virtually localized using a Head-Related Transfer Function (HRTF) of the user toward the user, acquiring a second position of the space around the user, the second position being estimated by the user who has listened to the reproduced sound data as a position of the sound source; and authenticating the user according to a coincidence between the first position and the second position. 1. A user authentication method , comprising: the sound source is in a first position of a space around a user,', 'the sound source is virtually localized, based on a Head-Related Transfer Function (HRTF) of the user convoluted with the sound data, and', 'the sound source is virtually localized by use of at least one of an earphone, a headphone, or a transaural system;, 'reproducing sound data of a sound source, wherein'}acquiring, as a user input, a second position of the space around the user, wherein the user input is based on the reproduced sound data; andauthenticating the user based on at least one coincidence between the first position and the second position.2. The user authentication method according to claim 1 , further comprising acquiring the user input from a terminal device of the user.3. The user authentication method according to claim 1 , wherein the sound data is reproduced binaurally.4. The user authentication method according to claim 1 , wherein the first position is from a position group in the space around the user.5. The user authentication method according to claim 4 , whereinthe position group includes a plurality of positions, andeach position of the plurality of positions has a different height.6. The user authentication method according to claim 4 , wherein the first position is in front of the user.7. The user authentication method according to claim 4 , wherein the position group further includes the second ...

Methods for Audio Signal Transient Detection and Decorrelation Control

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

REDUCING CORRELATION BETWEEN HIGHER ORDER AMBISONIC (HOA) BACKGROUND CHANNELS

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

RENDERING OF AUDIO OBJECTS WITH APPARENT SIZE TO ARBITRARY LOUDSPEAKER LAYOUTS

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

AUDIO PROVIDING APPARATUS AND AUDIO PROVIDING METHOD

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

USER CUSTOMIZABLE HEADPHONE SYSTEM

An apparatus may include a first headphone. The first headphone may include a first audio transducer and a second audio transducer. The first audio transducer may generate sound according to a first spectral mask and the second audio transducer may generate sound according to a second spectral mask. The first spectral mask and the second spectral mask may be adjustable at the first headphone. The first headphone may include a digital interface. The apparatus may further include a headphone controller to control the first headphone. The headphone controller may receive an audio signal from a portable electronic device and/or the headphone controller may transmit digital information representing speech, sound, or music to the first headphone. In response to a user input, the headphone controller may cause an adjustment to one or more of the first spectral mask or the second spectral mask. 1. A headphone apparatus comprising:a first headphone including a first audio transducer and a second audio transducer, wherein the first audio transducer generates sound according to a first spectral mask and the second audio transducer generates sound according to a second spectral mask, wherein the first spectral mask and the second spectral mask are adjustable at the first headphone, and wherein the first headphone includes a digital interface that receives audio information and audio reproduction control information; anda headphone controller to control the first headphone, wherein the headphone controller receives an audio signal from a portable electronic device, wherein the headphone controller transmits digital information including the audio information and the audio reproduction control information to the first headphone, wherein the headphone controller, in response to a user input, generates the audio reproduction control information that causes an adjustment to one or more of the first spectral mask or the second spectral mask.2. The headphone apparatus according to ...

SIMULATING ACOUSTIC OUTPUT AT A LOCATION CORRESPONDING TO SOURCE POSITION DATA

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

DECODING OF AUDIO SCENES

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

Manipulation of Playback Device Response Using Signal Processing

An example playback device receives left and right channels of audio content and generates a center channel of the audio content by combining at least a portion of the left right channels. The playback device generates first and second side channels of the audio content by combining the center channel and a difference of the left channel and the right channel and combining the center channel and an inverse of the difference of the left channel and the right channel, respectively. The playback device plays back the center channel of the audio content according to a first radiation pattern having a maximum aligned with a first direction, the first side channel according to a second radiation pattern having a maximum aligned with a second direction, and the second side channel according to a third radiation pattern having a maximum aligned with a fourth direction.

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

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

SOUND WALL

The invention relates to a sound wall comprising in total M elements comprising a front face (F), where the sound wall () has a lateral extension along an x-axis and a vertical extension along a y-axis, where the sound wall () comprises N elements (), where N≤M, which N elements in the front face (F) of each respective of these N elements are provided with a sound radiating unit, such as one or more loudspeaker units, configured to radiate sound energy from the respective front face and into the surroundings, such that a sound field can be created in front of the sound wall; where the sound wall is provided with signal providing means configured to provide the sound radiating units with a signal that is a combination of input signals, such as the left and right channel signals S(t), S(t) of a stereophonic signal. In embodiments of the invention, the sound wall () further comprises A elements (), where A≤M−N, which A elements () are provided with sound absorbing means associated with the front face (F), whereby the sound wall () can be used to reduce the reverberation time of a room or other enclosure, in which the sound wall () is provided and/or H elements (), where H≤M−N−A, which H elements () comprises hardware components configured to control and provide signals to the sound radiating units in said N elements (). The invention also relates to a method for controlling a sound field in front of the sound wall. 111133. A sound wall () comprising in total M elements comprising a front face (F) , where the sound wall () has a lateral extension along an x-axis and a vertical extension along a y-axis , where the sound wall () comprises N elements () , where N≤M; which N elements in the front face (F) of each respective of these N elements () are provided with a sound radiating unit , such as one or more loudspeaker units , configured to radiate sound energy from the respective front face and into the surroundings , such that a sound field can be created in front of the ...

Dynamic Audio Upmixer Parameters For Simulating Natural Spatial Variations

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

System and Method for Adaptive Audio Signal Generation, Coding and Rendering

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

Method and apparatus for facilitating spatial audio capture with multiple devices

A method, apparatus and computer program product are provided in order to facilitate spatial audio capture by utilizing multiple audio recording devices. In the context of a method, spatial characteristics of sound recorded by a first device are determined. The method also mapping, with a processor, sound recorded by a second device to a location and orientation of the first device. As such, the spatial characteristics of the sound recorded by the second device are modified to match the spatial characteristics of the sound recorded by the first device.

SYSTEM AND METHOD FOR EVALUATING MEDIA SEGMENTS FOR INTERESTINGNESS

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

Method and apparatus for an ultrasonic emitter system floor audio unit

Methods and systems are provided for audio devices with enhanced directional operations. In a system that includes one or more audio output components, positioning information associated with at least a part of a user's body maybe determined, the positioning information including at least orientation related information. The one or more audio output components may be controlled based on the determined positioning information, with the determined positioning information corresponding to a positional nature of the user within a three-dimensional space around the user, and where the controlling includes providing a three-dimensional audio environment according to the user.

AUDIO PROCESSING APPARATUS WITH CHANNEL REMAPPER AND OBJECT RENDERER

Embodiments are described for rendering spatial audio content through a system that is configured to reflect audio off of one or more surfaces of a listening environment. The system includes an array of audio drivers distributed around a room, wherein at least one driver of the array of drivers is configured to project sound waves toward one or more surfaces of the listening environment for reflection to a listening area within the listening environment and a renderer configured to receive and process audio streams and one or more metadata sets that are associated with each of the audio streams and that specify a playback location in the listening environment. 1. An audio processing apparatus comprising:an input interface for receiving channel metadata, object metadata, multichannel audio signals, and object audio signals;an audio decoder for decoding the multichannel audio signals to produce decoded multichannel audio signals and for decoding the object audio signals to produce decoded object audio signals;a channel remapper for converting the decoded multichannel audio signals from a transmitted channel configuration to a desired reproduction configuration based on the channel metadata to produce converted multichannel audio signals;an object renderer for rendering the decoded object audio signals based on the object metadata to produce rendered object audio signals, the object metadata comprising a set of parameters, wherein the set of parameters includes spatial position and size; andan output interface for outputting the converted multichannel audio signals or rendered object audio signals.2. The audio processing apparatus of wherein the spatial position is defined by one or more metadata parameters of the object metadata using a spherical coordinate system.3. The audio processing apparatus of wherein the object metadata further includes a parameter representing a width of an audio object.4. The audio processing apparatus of wherein the object metadata further ...

DECODING OF AUDIO SCENES

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

METHOD FOR PROCESSING OF SOUND SIGNALS

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

VIRTUAL SIMULATION OF SPATIAL AUDIO CHARACTERISTICS

Embodiments of the present invention are directed to a system and method for demonstrating spatial performance of a demonstration speaker model to consumers in order to evaluate different speakers. The system and method comprise a microphone array for recording the output of the demonstration speaker model. The system and method comprise acoustic input samples for processing to an acoustic output and a processor for determining characteristics of each microphone recording, and processing an acoustic input sample and characteristics of each microphone recording corresponding to a selected demonstration speaker model. The system and method further comprise a reference speaker model for outputting an acoustic signal based on the result of the processing. The processing compensates for the performance characteristic of the reference speaker and the performance characteristic of the selected demonstration speaker so as to mimic the spatial characteristics of the demonstration speaker while avoiding bias from the reference speaker. 1. A system for simulating acoustic performance of speakers , comprising:at least one reference speaker;a user interface;storage medium storing electronic data representing spatial audio characteristics of at least one demonstration speaker including a directivity pattern reflecting acoustic responses of the at least one demonstration speaker measured at multiple points in space; and receive, via said user interface, a first user selection of a first demonstration speaker;', 'retrieve, from said storage medium, electronic data representing spatial audio characteristics of said first demonstration speaker;', 'simulate acoustic performance of said first demonstration speaker by applying said spatial audio characteristics of said first demonstration speaker to an acoustic sample to generate a first simulated acoustic output; and', 'output said first simulated acoustic output via said at least one reference speaker., 'a computer processor or ...

Audio decoder for audio channel reconstruction

A method and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.

METHOD AND APPARATUS FOR PROCESSING MULTIMEDIA SIGNALS

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. 1. (canceled)2. A method for processing a multimedia signal , the method comprising:receiving a multimedia signal;receiving a set of filter coefficients for each subband, wherein the set of filter coefficients is truncated frequency-dependently from a set of proto-type subband filter coefficients based on a filter order for a corresponding subband, and wherein the filter order determines a length of the set of filter coefficients for each subband and is determined to be variable in a frequency domain; andfiltering each subband signal of the multimedia signal by using the set of filter coefficients corresponding thereto.3. The method of claim 2 , wherein the filter order is determined based at least part on energy decay time information of the corresponding subband.4. The method of claim 3 , wherein the energy decay time information is obtained from one or more sets of proto-type subband filter coefficients for the corresponding subband.5. The method of claim 3 , wherein the multimedia signal includes an audio signal claim 3 , andwherein the set of proto-type subband filter coefficients is a set of binaural room impulse response (BRIR) filter coefficients in the frequency domain.6. The method of claim 5 , wherein the energy decay time information includes reverberation time information.7. The method of claim 2 , wherein the filter order has a single value for each subband.8. An apparatus for processing a multimedia signal claim 2 , the apparatus comprising: receive a multimedia signal;', 'receive a set of filter coefficients for each subband, wherein the set of filter coefficients is truncated frequency-dependently from a ...

REAL-TIME IMMERSIVE MEDIATED REALITY EXPERIENCES

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

METHOD FOR AUDIO REPRODUCTION IN A MULTI-CHANNEL SOUND SYSTEM

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

APPARATUS AND METHOD FOR PROCESSING MULTI-CHANNEL AUDIO SIGNAL

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

REAL-TIME IMMERSIVE MEDIATED REALITY EXPERIENCES

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

AUDIO PROCESSING APPARATUS

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

BINAURAL RENDERING METHOD AND APPARATUS FOR DECODING MULTI CHANNEL AUDIO

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

SPEAKER FOR REFLECTING SOUND OFF VIEWING SCREEN OR DISPLAY SURFACE

Embodiments are described for rendering spatial audio content through a system that is configured to reflect audio off of one or more surfaces of a listening environment. The system includes an array of audio drivers distributed around a room, wherein at least one driver of the array of drivers is configured to project sound waves toward one or more surfaces of the listening environment for reflection to a listening area within the listening environment and a renderer configured to receive and process audio streams and one or more metadata sets that are associated with each of the audio streams and that specify a playback location in the listening environment. 1. A speaker for generating sound waves in a listening environment , the speaker comprising:a speaker cabinet; andan audio driver enclosed in or coupled to the speaker cabinet, wherein the audio driver is configured to project sound waves toward one or more surfaces of the listening environment for reflection to a listening area within the listening environment, wherein the one or more surfaces include a viewing screen or display surface located at a front of the listening environment, and wherein the front of the listening environment is a direction facing a listener in the listening environment.2. The speaker of wherein the audio driver is a front-firing claim 1 , side-firing claim 1 , upward firing claim 1 , or downward firing driver.3. The speaker of wherein the audio driver is a full range speaker claim 1 , a subwoofer claim 1 , or a tweeter.4. The speaker of wherein the viewing screen or display surface includes a relatively flat surface.5. The speaker of wherein the listening environment includes a ceiling claim 1 , a floor claim 1 , a front wall claim 1 , a back wall claim 1 , and side walls claim 1 , wherein the front wall is positioned facing the listener in the listening environment.6. The speaker of wherein the screen or the display surface is located substantially on or near the front wall.7. The ...

METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR CONDUCTING INTERACTIVE SOUND PROPAGATION AND RENDERING FOR A PLURALITY OF SOUND SOURCES IN A VIRTUAL ENVIRONMENT SCENE

Methods, systems, and computer readable media for conducting interactive sound propagation and rending for a plurality of sound sources in a virtual environment scene are disclosed. According to one method, the method includes decomposing a virtual environment scene containing a plurality of sound sources into a plurality of partitions and forming a plurality of source group clusters, wherein each of the source group clusters includes two or more of the sound sources located within a common partition. The method further includes determining, for each of the source group clusters, a single set of sound propagation paths relative to a listener position and generating a simulated output sound at a listener position using sound intensities associated with the determined sets of sound propagation paths. 1. A method for conducting interactive sound propagation and rending for a plurality of sound sources in a virtual environment scene , the method comprising:decomposing a virtual environment scene containing a plurality of sound sources into a plurality of partitions;forming a plurality of source group clusters, wherein each of the source group clusters includes two or more of the sound sources located within a common partition;determining, for each of the source group clusters, a single set of sound propagation paths relative to a listener position; andgenerating a simulated output sound at a listener position using sound intensities associated with the determined sets of sound propagation paths.2. The method of wherein generating a simulated output sound includes sorting each of the sound propagation paths based on an amount of Doppler shifting exhibited by the sound propagation path.3. The method of comprising rendering a sound intensity using fractional delay line interpolation on a first group of the sound propagation paths that exhibits an amount of Doppler shifting that exceeds a predefined threshold.4. The method of comprising rendering a sound intensity using a ...

Audio signal decoder, audio signal encoder, method for providing an upmix signal representation, method for providing a downmix signal representation, computer program and bitstream using a common inter-object-correlation parameter value

An audio signal decoder for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information and in dependence on a rendering information has an object parameter determinator. The object parameter determinator is configured to obtain inter-object-correlation values for a plurality of pairs of audio objects. The object parameter determinator is configured to evaluate a bitstream signaling parameter in order to decide whether to evaluate individual inter-object-correlation bitstream parameter values to obtain inter-object-correlation values for a plurality of pairs of related audio objects, or to obtain inter-object-correlation values for a plurality of pairs of related audio objects using a common inter-object-correlation bitstream parameter value. The audio signal decoder also has a signal processor configured to obtain the upmix signal representation on the basis of the downmix signal representation and using the inter-object-correlation values for a plurality of pairs of related objects and the rendering information.

INTERNET AUDIO SOFTWARE METHOD

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

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

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

System and Tools for Enhanced 3D Audio Authoring and Rendering

Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment. 1. A method , comprising:receiving audio reproduction data comprising one or more audio objects and metadata associated with each of the one or more audio objects;receiving reproduction environment data comprising an indication of a number of reproduction speakers in the reproduction environment and an indication of the location of each reproduction speaker within the reproduction environment; andrendering the audio objects into one or more speaker feed signals by applying an amplitude panning process to each audio object, wherein the amplitude panning process is based, at least in part, on the metadata associated with each audio object and the location of each reproduction speaker within the reproduction environment, and wherein each speaker feed signal corresponds to at least one of the reproduction speakers within the reproduction environment;wherein the metadata associated with each audio object includes audio object coordinates indicating the intended reproduction position of the audio object within the reproduction environment and a snap flag indicating whether the amplitude panning process should render the audio object into a single speaker feed signal or apply panning rules to render the audio object into a plurality of speaker feed signals.2. The method of claim 1 , wherein:the snap flag indicates the amplitude panning process should render the audio object into a single speaker feed signal; andthe amplitude panning process renders the audio object into a ...

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

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

Reverberation Generation for Headphone Virtualization

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

Audio System with Binaural Elements and Method of Use with Perspective Switching

An audio system with binaural elements with a computer processor connected to data storage. The audio system is capable of perspective switching between different binaural elements present in an audio recording. 1. A system for playback of an audio recording with a binaural audio recording , comprising: a binaural audio recording;', 'a menu selection button to select features from the menu listing;', 'a menu list with several features;'}], 'an audio recording comprisinga data storage configured to store at least one audio recording having the binaural audio recording and to present a series of audio recording choices to a listener, the binaural audio recording comprising a member, or combination of members, from the group consisting of: holophonic sound effects, a dialog, music, narration, machinery, environmental sounds, and/or script; a computer processor connected to the data storage, the computer processor configured to receive and playback the audio recording as an audio output signal;', 'wearable headgear in communication with the computer processor configured to receive and play the audio output signal through a first output and a second output;', [ computer instructions for instructing the computer processor to present the series of audio recording choices to the listener and enable the listener to select one of the audio recordings from a series of audio recording choices using the listener personal computing device, each audio recording having the binaural audio recording;', 'a perspective switching button viewable, physically available, or audio activated, on the display and connected to computer instructions configured to instruct the personal computing device processor and/or computer processor to present non-selected individual audio sources to the listener for selection at any time during the playback; and, 'said data storage further comprising, 'a plurality of navigation buttons viewable, physically available, or audio activated, on the display and ...

SIMULATING ACOUSTIC OUTPUT AT A LOCATION CORRESPONDING TO SOURCE POSITION DATA

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

METHODS AND SYSTEMS FOR PROVIDING VIRTUAL SURROUND SOUND ON HEADPHONES

Method and system for providing virtual surround sound on headphones using input audio. Embodiments herein relate to sound processing and more particularly to providing surround sound on headphones. Embodiments herein disclose a method and system for simulating surround sound on a headphone, by emulating multiple speakers in 3D space by processing audio using Head Related Transfer Function (HRTF) filters and other audio processing filters, wherein the input to the headphone is stereo input. 1301. A method for simulating surround sound on a headphone () , the method comprising{'b': '302', 'filtering an audio input to extract a plurality of elements present in the audio input by a plurality of audio filters ();'}{'b': '303', 'applying spatial cues to each of the extracted plurality of elements by a plurality of Head Related Transfer Function (HRTF) filters () for a determined layout of a plurality of virtual speakers;'}{'b': 303', '304, 'tuning output of the plurality of HRTF filters () by a plurality of tuning engines ();'}{'b': 304', '305, 'mixing the tuned output of the plurality of tuning engines () by a three-dimensional (3D) audio mixer (); and'}{'b': '301', 'rendering the mixed tuned output on a left and right channel of the headphone ().'}2. The method claim 1 , as claimed in claim 1 , wherein the audio input is at least one of a stereo input; and a multi-channel audio input.3. The method claim 1 , as claimed in claim 1 , wherein filtering the audio input to extract a plurality of elements further comprises{'b': 302', '1', '302', '2, 'filtering frequency components from the audio input to which a human auditory system is sensitive by the audio filters (-, -);'}{'b': 302', '3', '302', '4, 'filtering lower-mid frequency components from the audio input by the audio filters (-, -);'}{'b': 302', '5, 'filtering high frequency components from the audio input by the audio filter (-); and'}{'b': 302', '6, 'filtering low frequency components from the audio input by the ...

SPATIAL AUDIO SYSTEM FOR PLAYING LOCATION-AWARE DYNAMIC CONTENT

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

AV Timing Measurement and Correction for Digital Television

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

Audio apparatus and audio providing method thereof

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

METHOD FOR TRANSMITTING AND RECEIVING AUDIO DATA AND APPARATUS THEREFOR

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

System and Tools for Enhanced 3D Audio Authoring and Rendering

Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment. 1. A method , comprising:receiving audio reproduction data comprising one or more audio objects and metadata associated with each of the one or more audio objects;receiving reproduction environment data comprising an indication of a number of reproduction speakers in the reproduction environment and an indication of the location of each reproduction speaker within the reproduction environment; andrendering the audio objects into one or more speaker feed signals by applying an amplitude panning process to each audio object, wherein the amplitude panning process is based, at least in part, on the metadata associated with each audio object and the location of each reproduction speaker within the reproduction environment, and wherein each speaker feed signal corresponds to at least one of the reproduction speakers within the reproduction environment;wherein the metadata associated with each audio object includes audio object coordinates indicating the intended reproduction position of the audio object within the reproduction environment and metadata indicating audio object spreads in two or more of three dimensions, wherein the audio object spreads are the same in the two or more dimensions, and wherein the rendering involves controlling the audio object spreads in the two or more dimensions in response to the metadata.2. An apparatus , comprising:an interface system; and receiving, via the interface system, audio reproduction data comprising one or more audio objects and ...

AUDIO SIGNAL PROCESSING METHOD

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

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

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

Method for audio reproduction in a multi-channel sound system

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

RENDERER CONTROLLED SPATIAL UPMIX

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

MULTIDIMENSIONAL VIRTUAL LEARNING SYSTEM AND METHOD

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

Multidimensional virtual learning system and method

A process and system for generating three dimensional sound conferencing includes generating a virtual map with a plurality of positions, each participant selecting one of the positions, determining a direction from each position to each other position on the map, determining a distance from each position to each other position on the map, receiving sound from each participant, mixing the received sound, transforming the mixed sound into binaural audio, and directing the binaural audio sound to each participant via a speaker associated with the virtual position of the speaking participant. The result is a clarified sound that gives to the listening participant a sense of where the speaking participant is positioned relative to the listening participant.

Reverberation Generation for Headphone Virtualization

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

UPMIXING OF AUDIO SIGNALS

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

VEHICLE AUDIO SYSTEM WITH REVERBERANT CONTENT PRESENTATION

Various implementations include vehicle audio systems and related control methods. In some particular implementations, an audio system is configured to present audio content to a vehicle occupant to modify the perceived size of the vehicle cabin relative to its actual physical size. 1. An audio system for a vehicle , the audio system comprising:a plurality of speakers located in the vehicle;at least one transducer affixed to an overhead structure of the vehicle; andan amplifier connected with the plurality of speakers and the at least one transducer, the amplifier comprising circuitry configured to process entertainment audio signals to generate spatially enhanced entertainment audio content that is provided to the plurality of speakers for output, and reverberant content that is provided to the at least one transducer for output,wherein presentation of both the spatially enhanced entertainment audio content and the reverberant content within a cabin of the vehicle causes an occupant of the vehicle to have a first auditory perception of a size of the cabin, while presentation of only the spatially enhanced entertainment audio content within the cabin causes the occupant to have a second, different auditory perception of the size of the cabin.2. The system of claim 1 , wherein the first auditory perception of the size of the cabin is larger than the second auditory perception of the size of the cabin.3. The system of claim 1 , wherein the overhead structure comprises a headliner of the vehicle claim 1 , and wherein the plurality of speakers are located in regions of the vehicle other than the headliner.4. The system of claim 1 , wherein the at least one transducer comprises an exciter.5. The system of claim 1 , wherein the at least one transducer comprises a speaker distinct from the plurality of speakers.6. The system of claim 1 , wherein the circuitry in the amplifier comprises first circuitry for applying spatial enhancement processing to the entertainment audio ...

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

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

Manipulation of Playback Device Response Using Signal Processing

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

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

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

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

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

APPARATUS AND METHOD FOR SEPARATING MULTI-CHANNEL AUDIO SIGNAL

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

AUDIO RENDERING SYSTEM CATEGORISING GEOSPATIAL OBJECTS

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

METHOD AND APPARATUS FOR REPRODUCING THREE-DIMENSIONAL AUDIO

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

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

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

SYSTEM AND PROGRAM FOR IMPLEMENTING THREE-DIMENSIONAL AUGMENTED REALITY SOUND BASED ON REALISTIC SOUND

Disclosed is an augmented reality sound implementation system for executing an augmented reality sound implementation method. The system includes a first computing device of a first user; and a first sound device which is worn by the first user so that the first user can receive a three-dimensional augmented reality sound, is connected to the first computing device in a wired or wireless manner, and includes a sound recording function. 1. An augmented reality sound implementation system for performing a method for an augmented reality sound , the system comprising:a first computing device of a first user; anda first sound device which is worn by the first user such that the first user can receive a three-dimensional augmented reality sound, is connected to the first computing device in a wired or wireless manner, and includes a sound recording function,wherein the method comprising:obtaining, by the first sound device, realistic sound information, which indicates a realistic sound, to transmit the realistic sound information to the first computing device;obtaining, by the first computing device, a first virtual sound which indicates a sound generated from a virtual reality game executed by the first computing device;generating, by the first computing device, a three-dimensional augmented reality sound based on the realistic sound information and the first virtual sound; andproviding, by the first computing device, the three-dimensional augmented reality sound to the first user through the first sound device.2. The system of claim 1 , wherein the method further comprising:obtaining, by the first computing device, direction feature information indicating a location where the realistic sound is generated; andfurther considering, by the first computing device, the direction feature information to generate the three-dimensional augmented reality sound.3. The system of claim 2 , wherein the first computing device determines whether the first user and a second user spaced ...

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

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

Location information through directional sound provided by mobile computing device

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

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

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

APPARATUS AND METHOD FOR EDGE FADING AMPLITUDE PANNING

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

STEREO EXTENSION APPARATUS AND METHOD

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

Audio Processing Method and Audio Processing Apparatus

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

MULTIMEDIA AUDIO INSERTER

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

SOUND PROCESSING APPARATUS AND METHOD, AND PROGRAM

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

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

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

SYSTEM AND METHOD FOR ENHANCING VIRTUAL AUDIO HEIGHT PERCEPTION

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

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

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

SPATIALLY REPRESENTING GRAPHICAL INTERFACE ELEMENTS AS BINAURAL AUDIO CONTENT

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

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

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

Method for Playing Back the Sound of a Digital Audio Signal

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

SOUND PROCESSING APPARATUS AND SOUND PROCESSING SYSTEM

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

AUDIO SIGNAL PROCESSING METHOD

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

System and Tools for Enhanced 3D Audio Authoring and Rendering

Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment. 1. A method , comprising:receiving audio reproduction data comprising one or more audio objects and metadata associated with each of the one or more audio objects;receiving reproduction environment data comprising an indication of a number of reproduction speakers in the reproduction environment and an indication of the location of each reproduction speaker within the reproduction environment; andrendering the audio objects into one or more speaker feed signals by applying an amplitude panning process to each audio object, wherein the amplitude panning process is based, at least in part, on the metadata associated with each audio object and the location of each reproduction speaker within the reproduction environment, and wherein each speaker feed signal corresponds to at least one of the reproduction speakers within the reproduction environment;wherein the metadata associated with each audio object includes audio object coordinates indicating the intended reproduction position of the audio object within the reproduction environment and metadata indicating an audio object spread in one or more of three dimensions, and wherein the rendering involves controlling the audio object spread in the one or more dimensions in response to the metadata.2. The method of claim 1 , wherein the metadata indicates audio object spreads in two or more of the three dimensions claim 1 , and wherein the rendering involves controlling the audio object spread in the two or more dimensions in ...

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

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

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

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

SOUND PROCESSING APPARATUS AND SOUND PROCESSING SYSTEM

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