Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by a new type of decode matrix for sound field data, and a new way to obtain the decode matrix.
Abstract: The present invention provides a method of processing a source left view and a source right view of a 3D image comprising the steps of: identifying at least one ghosted region or ghosting region in the left view and/or in the right view; processing the identified regions in order to create a binocular suppression effect for this identified regions then providing a new 3D image formed by the processed left and/or right views.
Abstract: A system and method are described for transforming stereo signals into mid and side components xm and xs to apply processing to only the side-component xs and avoid processing the mid-component. By avoiding alteration to the mid-component XM, the system and method may reduce the effects of ill-conditioning, such as coloration that may be caused by processing a problematic mid-component xM while still performing crosstalk cancellation and/or generating virtual sound sources. Additional processing may be separately applied to the mid and side components xM and xs and/or particular frequency bands of the original stereo signals to further reduce ill-conditioning.
Type:
Grant
Filed:
September 29, 2015
Date of Patent:
August 28, 2018
Assignee:
Apple Inc.
Inventors:
Martin E. Johnson, Sylvain J. Choisel, Daniel K. Boothe, Mitchell R. Lerner
Abstract: 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.
Abstract: 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.
Type:
Grant
Filed:
August 24, 2017
Date of Patent:
August 21, 2018
Assignee:
SAMSUNG ELECTRONICS CO., LTD.
Inventors:
Sang-bae Chon, Sun-min Kim, Jae-ha Park, Sang-mo Son, Hyun Jo, Hyun-joo Chung
Abstract: A method and an electronic device for outputting an audio signal in the electronic device is provided. The electronic device includes a first speaker, a second speaker, and an audio processor that creates, from an audio signal, a first frequency audio signal corresponding to a first frequency band by using a low pass filter, synthesizes the created first frequency audio signal and the audio signal to create a synthetic audio signal, creates, from the synthetic audio signal, a second frequency audio signal corresponding to a second frequency band by using a high pass filter, outputs the created second frequency audio signal through the first speaker, and outputs the created synthetic audio signal through the second speaker.
Type:
Grant
Filed:
December 15, 2016
Date of Patent:
August 14, 2018
Assignee:
Samsung Electronics Co., Ltd
Inventors:
Taiyong Kim, Dongeon Kim, Seungsoo Nam, Juhee Jang, Jeok Lee, Jaehyun Kim, Hochul Hwang
Abstract: Technology for adjusting a receiver timing of a wireless device in a Coordinated MultiPoint (CoMP) system is disclosed. One method can include the wireless device receiving a plurality of node specific reference signals (RSs) from a plurality of cooperating nodes in a coordination set of the CoMP system. The coordination set includes at least two cooperating nodes. The wireless device can estimate a composite received RS timing from a plurality of received RS timings generated from the plurality of node specific RSs. The received RS timings represent timings from the at least two cooperating nodes. The wireless device can adjust the receiver timing based on the composite received RS timing. A node specific RS can include a channel-state information reference signal (CSI-RS).
Abstract: Disclosed is a speaker diarization process for determining which speaker is speaking at what time during the course of a conversation. The entire process can be most easily described in five main parts: Segmentation where speech/non-speech decisions are made; frame feature extraction where useful information is obtained from the frames; segment modeling where the information from the frame feature extraction is combined with segment start and end time information to create segment specific features; speaker decisions when the segments are clustered to create speaker models; and corrections where frame level corrections are applied to the information extracted.
Type:
Grant
Filed:
May 3, 2016
Date of Patent:
July 17, 2018
Assignee:
SESTEK Ses velletisim Bilgisayar Tekn. San. Ve Tic A.S.
Inventors:
Mustafa Levent Arslan, Mustafa Erden, Sedat Demirba{hacek over (g)}, Gökçe Sarar
Abstract: An audio signal processing device for downmixing of a first input signal and a second input signal to a downmix signal having: a dissimilarity extractor configured to receive the first input signal and the second input signal as well as to output an extracted signal, which is lesser correlated with respect to the first input signal than the second input signal and a combiner configured to combine the first input signal and the extracted signal in order to obtain the downmix signal.
Type:
Grant
Filed:
March 25, 2016
Date of Patent:
July 10, 2018
Assignee:
Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
Inventors:
Alexander Adami, Emanuel Habets, Juergen Herre
Abstract: An arrangement, for reproducing audio data of an acoustic scene, adapted for generating audio signals for at least a first and a second headphone channel of a headphone assembly, the audio signals corresponding to at least one audio object and/or sound source in the acoustic scene comprising at least one given close range and at least one given distant range arranged around a listener, the arrangement comprising a first headphone channel; a second headphone channel; a basic channel provider comprising at least a basic system adapted for reproducing audio signals corresponding to at least one audio object and/or sound source arranged in at least one distant range; a proximity channel provider comprising at least a proximity system adapted for reproducing audio signals corresponding to at least one audio object and/or sound source arranged in at least one close range.
Type:
Grant
Filed:
May 23, 2014
Date of Patent:
July 10, 2018
Assignee:
BARCO NV
Inventors:
Markus Mehnert, Robert Steffens, Martin Dausel, Henri Meissner
Abstract: This application describes methods of signal processing and spatial audio synthesis. One such method includes accepting an auditory signal and generating an impression of auditory virtual reality by processing the auditory signal to impute a spatial characteristic on it via convolution with a plurality of head-related impulse responses. The processing is performed in a series of steps, the steps including: performing a first convolution of an auditory signal with a characteristic-independent, mixed-sign filter and performing a second convolution of the result of first convolution with a characteristic-dependent, sparse, non-negative filter. In some described methods, the first convolution can be pre-computed and the second convolution can be performed in real-time, thereby resulting in a reduction of computational complexity in said methods of signal processing and spatial audio synthesis.
Type:
Grant
Filed:
June 8, 2015
Date of Patent:
July 3, 2018
Assignee:
University of Maryland, College Park
Inventors:
Yuancheng Luo, Ramani Duraiswami, Dmitry N. Zotkin
Abstract: Apparatus and methods for reducing crosstalk in personal audio equipment are provided. In an example, a method to reduce headset audio crosstalk can include applying a first signal to a first speaker channel of a headset, coupling a second speaker channel to a first input of a comparator of a crosstalk compensation circuit using a first switch of the crosstalk compensation circuit, the switch and detect circuit including the crosstalk compensation circuit, coupling a first resistor divider to a second input of the comparator using a second switch of the crosstalk compensation circuit, and adjusting a resistance setting of the first resistor divider from an initial setting using an output of the comparator.
Abstract: The audio amplifier includes a variable gain amplifier receiving the input audio signal and providing the output signal, whereby the output signal corresponds to the input signal amplified by a limiter gain. The audio amplifier further includes a limiter gain calculation unit, thus the input signal is amplified by the limiter gain. A control unit receives a signal representative of the input signal and is configured to estimate, based on a mathematical model, the input current or the total output current of the audio amplifier thus providing an estimated current signal corresponding to (and resulting from) the output signal, whereby the limiter gain calculation unit is configured to calculate, dependent on the estimation, the limiter gain such that the actual input current or the total output current of the audio amplifier does not exceed a threshold current value.
Type:
Grant
Filed:
May 17, 2016
Date of Patent:
June 26, 2018
Assignee:
Harman Becker Automotive Systems GmbH
Inventors:
Gerhard Pfaffinger, Franz Lorenz, Stefan Zuckmantel, Markus Christoph
Abstract: A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.
Abstract: The present technology relates to an acoustic signal processing device, an acoustic signal processing method, and a program capable of improving a feeling of localization of a sound image at a position deviated leftward or rightward from a median plane of a listener. A transaural processing unit performs a predetermined transaural process for an input signal by using a sound source opposite side HRTF and a sound source side HRTF to generate a first acoustic signal, and a second acoustic signal containing attenuated components in a first band which is the lowest band, and a second band which is the second lowest band in a range of a predetermined first frequency or higher frequencies, in bands of appearance of notches in the sound source opposite side HRTF. A subsidiary signal synthesis unit adds a first subsidiary signal constituted by a component in a predetermined band of the second acoustic signal to the first acoustic signal to generate a third acoustic signal.
Abstract: A method of interpolating a head-related transfer function (HRTF) and an audio output apparatus using the same are disclosed. The method includes receiving HRTF data corresponding to a point at which an altitude angle and an azimuth angle cross and receiving complementary information about a point at which the HRTF data is present, generating an HRTF interpolation signal corresponding to an altitude angle of a sound localization point, using HRTF data corresponding to two points constituting an altitude angle segment nearest the sound location point, calculating an amount of variation up to an azimuth angle ? of the sound localization point, using complementary information of two points constituting an azimuth angle segment nearest the sound localization point, and generating a final HRTF interpolation signal corresponding to the sound localization point by applying the amount of variation to the HRTF interpolation signal corresponding to the altitude angle of the sound localization point.
Abstract: The present disclosure is directed towards systems, methods and devices for tamper proofing documents by embedding data in a biometric identifier. The biometric identifier may be a real or synthetic fingerprint. In one non-limiting exemplary embodiment, a method for embedding data in a biometric identifier includes determining data to embed in the biometric identifier, determining at least one control parameter, and generating at least one biometric identifier feature. The method further includes including the at least one generated biometric identifier feature in the biometric identifier such that the determined data is embedded in the biometric feature. The determined data is based on a variable data stream. The control parameter controls the generation of at least one biometric identifier feature. The control parameter is based on the determined data. The generated biometric identifier feature is based on the determined control parameter.
Abstract: A directivity control method is provided for controlling a directivity of a sound collected by a first sound collector including a plurality of microphones. The directivity control method includes: forming a directivity of the sound in a direction toward a monitoring target corresponding to a first designated position in an image displayed on a display; obtaining information on a second designated position in the image displayed on the display, designated in accordance with a movement of the monitoring target, and changing the directivity of the sound toward the monitoring target corresponding to the second designated position by referring to the information on the second designated position.
Abstract: A recording method acquires, from a first musical instrument which outputs playing data which represents playing information, playing data of playing by the first musical instrument; generates a first waveform signal according to the played sounds of the first musical instrument which correspond to the playing data; generates a second waveform signal according to a sound including a sound emitted from the first musical instrument and other sounds; generates a third waveform signal wherein the first waveform signal is subtracted from the second waveform signal; generates audio data from the third waveform signal; and records the audio data.
Abstract: An object detection and notification system includes an object detection system including logic that detects presence of a target object within a sensing range of a sensor and determines a direction of the target object relative to the autonomous vehicle. A notification system includes logic that provides a sound cue using a speaker to an occupant of the autonomous vehicle that is indicative of location of the target object relative to the autonomous vehicle.
Type:
Grant
Filed:
July 18, 2016
Date of Patent:
May 1, 2018
Assignee:
Toyota Motor Engineering & Manufacturing North America, Inc.
Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by a new type of decode matrix for sound field data, and a new way to obtain the decode matrix.
Abstract: A sound reproduction system includes an electro-acoustic transducer and a transducer driver for driving the electro-acoustic transducer. The transducer drive includes a filter which is configured to reproduce at a listener's location an approximation to the local sound field that would be present at the listener's ears in recording space, taking into account the characteristics and intended position of the electro-acoustic transducer relative to the listener's ears. The electro-acoustic transducer includes a first sound emitter which provides an intermediate sound emission channel, and second and third sound emitters providing respective left and right sound emission channels. The first sound emitter is located intermediate of second and third sound emitters. Higher frequencies from at least one of the second and third sound emitters are transmitted closer to the first sound emitter while lower frequencies are transmitted away from the first sound emitter.
Abstract: A hearing aid system includes: a first hearing aid comprising a first set of microphones, a first beamformer, a first processing module, and a first receiver; and a second hearing aid comprising a second set of microphones, a second beamformer, a second processing module, and a second receiver; wherein the first beamformer in a first operating mode is configured to provide a first audio signal in accordance with a first primary spatial characteristic, wherein the second beamformer in the first operating mode is configured to provide a second audio signal in accordance with a second primary spatial characteristic, the first primary spatial characteristic having a first main lobe with a first direction, and the second primary spatial characteristic having a second main lobe with a second direction, wherein the second direction is different from the first direction.
Abstract: A driving device for a sound system by loudspeaker signals, wherein the sound system has a wave field synthesis loudspeaker array and one or several supply loudspeakers arranged separate from the wave field synthesis array includes an audio input for receiving at least one audio signal from at least one sound source, a position input for receiving information on a position of the sound source, a wave field synthesis unit for calculating loudspeaker signals for the loudspeakers of the wave field synthesis loudspeaker array, and a provider for providing the loudspeaker signal for the one or the several supply loudspeakers. The driving device enables a sound system by means of which sound localization becomes possible for the audience and at the same time pleasant levels can be achieved also in the first rows of the audience.
Type:
Grant
Filed:
March 18, 2016
Date of Patent:
April 24, 2018
Assignee:
Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
Abstract: 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.
Type:
Grant
Filed:
August 9, 2017
Date of Patent:
April 10, 2018
Assignee:
Dolby Laboraties Licensing Corporation
Inventors:
Charles Q. Robinson, Nicolas R. Tsingos, Christophe Chabanne
Abstract: A method is provided for optimizing acoustic localization at one or more listening positions in a listening environment such as, but not limited to, a vehicle passenger compartment. The method includes generating a sound field with a group of loudspeakers assigned to at least one of the listening positions, the group of loudspeakers including first and second loudspeakers, where each loudspeaker is connected to a respective audio channel; calculating filter coefficients for a phase equalization filter; configuring a phase response for the phase equalization filter such that binaural phase difference (??mn) at the at least one of the listening positions or a mean binaural phase difference (m??mn) averaged over the listening positions is reduced in a predefined frequency range; and filtering the audio channel connected to the second loudspeaker with the phase equalization filter.
Abstract: Method, equipment and apparatus for acquiring a spatial audio direction vector, the method including: determining a position of a sound source in a multi-sound system; setting a parameter comprising: a human response time ?t and a tolerance percentage ?; acquiring a sound signal from the sound source; and processing the sound signal by using the parameter and acquiring a corresponding spatial audio direction vector {right arrow over (E)} within each time interval ?t. A proportional constant D is determined according to a modulus of a spatial audio direction vector {right arrow over (E)}, and provides spatial information of depth for a virtual image corresponding to a multi-tone audio signal. A vector angle ?E the spatial audio direction vector {right arrow over (E)} provides spatial information of direction for the virtual image corresponding to the multi-tone audio signal, to improve viewer's viewing experience.
Type:
Grant
Filed:
July 22, 2016
Date of Patent:
March 13, 2018
Assignee:
Marvel Digital Limited
Inventors:
Ying Chiu Herbert Lee, Ho Sang Lam, Tin Wai Grace Li
Abstract: A method for compensating for acoustic crosstalk between a first and a second microphone unit being acoustically connected to a shared volume. The method includes the steps of providing a first output signal, Pout, from the first microphone unit, providing a second output signal, Uout, from the second microphone unit, and generating a compensated output signal by combining a portion of one of the output signals with the other output signal via addition or subtraction in order to compensate for acoustical crosstalk. The invention further relates to a microphone module configured to implement the before-mentioned method. The invention further relates to a hearing aid comprising the microphone module.
Abstract: A sound processing device includes a shift detection unit that detects a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and a correction unit that corrects a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point.
Abstract: A multichannel compensating audio system includes first and second compensation channels to psychoacoustically minimize deviations in a target response, to psychoacoustically move the physical position of a speaker and/or to psychoacoustically provide a substantially equal magnitude of sound from a plurality of speakers in a plurality of different listening positions. The first compensation channel may include a series connected delay circuit, a level adjuster circuit and a frequency equalizer circuit that generates a first compensated audio signal from a first audio signal. The second compensation channel may include a series connected delay circuit, a level adjuster circuit and a frequency equalizer circuit that generates a second compensated audio signal from a second audio signal. A first summing circuit is configured to receive at least the first audio signal and the second compensated audio signal and generate a first output signal for provision to a first speaker.
Type:
Grant
Filed:
April 16, 2015
Date of Patent:
February 6, 2018
Assignee:
Harman International Industries, Incorporated
Abstract: Provided are methods and apparatus for enhancing a signal-to-noise ratio. In an example, provided is an apparatus configured to modify audio to better match the way the human brain processes audio by modifying the audio to a form which takes advantage of human echolocation capabilities. When humans listen to audio, they subconsciously listen for an echo and thus subconsciously focus on listening to, and for, meaningful information in audio. The focus causes humans to ignore noise in the audio, which results in enhancing a signal-to-noise ratio.
Abstract: Systems and methods for monitoring permission to be in a location comprising: a secured area having at least one entry point with a mechanical gate having an open and closed position; at least two wireless proximity sensors attached to a portion of (or the area adjacent to) the mechanical gate; a token device in communication with the wireless proximity sensors that determine a location of the token device relative to one of the wireless proximity sensors to provide a detection data point and a set of detection data points for the group of detection data points; and a system computing device that calculates the shared proximity of the token device. If the token device contains a valid ticket and the shared proximity of the token device is within a predetermined area the system computing device will cause the mechanical gate to go to the open position.
Type:
Grant
Filed:
March 4, 2015
Date of Patent:
January 30, 2018
Inventors:
Micah Bergdale, Matthew Grasser, Nicholas Ihm, Gregory Valyer
Abstract: Examples disclosed herein relate to an electronic gaming device including one or more processors, a memory, and one or more display devices. The electronic gaming device further includes a left surround audio device, a right surround audio device, a dialog enhancing center channel speaker, a low frequency effects device, a left speaker, and a right speaker. The combination of the left surround audio device; the right surround audio device; the dialog enhancing center channel speaker; the low frequency effects device; the left speaker; and/or the right speaker produce one or more sound effects in a vertical direction.
Type:
Grant
Filed:
November 16, 2016
Date of Patent:
January 30, 2018
Assignee:
AGS LLC
Inventors:
Laura Elizabeth Taylor, Jason Dean Grace, Ian Robert Scott, Mark Andrew Thompson
Abstract: A sound field measuring device (1) obtains frequency characteristics by collecting output sound outputted from a pair of speakers (101a, 101b) installed at a narrow interval. A low-pass filter (22a) extracts low-range components of a first measurement signal. A high-pass filter (22b) extracts mid/high-range components of a second measurement signal different from the first measurement signal. A combined signal generation unit (22c) generates a combined signal by combining the low-range components of the first measurement signal and the mid/high-range components of the second measurement signal. An external output unit (6) outputs the first measurement signal to an audio system (102). A microphone (7) collects the first measurement signal and the combined signal simultaneously outputted from the pair of speakers. A Fourier transform unit (13) obtains the frequency characteristics of a sound field by Fourier transforming the signals collected.
Abstract: According to one aspect, an electronic device for detecting an audio accessory. The electronic device includes an audio jack having at least two detection terminals. The detection terminals are spaced apart and positioned within a socket of the audio jack so when an audio plug of the accessory is inserted into the socket of the audio jack, the detection terminals will be shorted. The presence of a short between the detection terminals is indicative that the audio accessory is present.
Type:
Grant
Filed:
June 29, 2016
Date of Patent:
January 23, 2018
Assignee:
BLACKBERRY LIMITED
Inventors:
Jens Kristian Poulsen, Yong Zhang, Per Magnus Fredrik Hansson
Abstract: Methods, apparatus, computer program products embodied on non-transitory computer-readable storage mediums, and systems of sound generation for monitoring user interfaces include storing a mapping of visual elements of a user interface (UI) to respective coordinates in a virtual space, each of the respective coordinates having a corresponding distance within the virtual space from a virtual listener. For each mapped visual element in a first set of mapped visual elements, a respective sound component is generated based on a state of the mapped visual element within the UI, and the distance between the coordinates of the mapped visual element and the virtual listener. A plurality of the sound components is then transmitted to a remote device.
Abstract: A method includes extracting a difference value through extraction of features of a front audio channel signal and a surround channel of multichannel sound content by setting the front audio channel signal and the surround channel as input and output channel signals, respectively, training a deep neural network (DNN) model by setting the input channel signal and the difference value as an input and an output of the DNN model, respectively, normalizing a frequency-domain signal of the input channel signal by converting the input channel signal into the frequency-domain signal, and extracting estimated difference values by decoding the normalized frequency-domain signal through the DNN model, deriving an estimated spectral amplitude of the surround channel based on the front audio channel signal and the difference value, and deriving an audio signal of a final surround channel by converting the estimated spectral amplitude of the surround channel into the time domain.
Type:
Grant
Filed:
November 18, 2016
Date of Patent:
January 9, 2018
Assignee:
GWANGJU INSTITUTE OF SCIENCE AND TECHNOLOGY
Inventors:
Hong Kook Kim, Su Yeon Park, Chan Jun Chun
Abstract: An audio enhancement system can provide spatial enhancement, low frequency enhancement, and/or high frequency enhancement for headphone audio. The spatial enhancement can increase the sense of spaciousness or stereo separation between left and right headphone channels. The low frequency enhancement can enhance bass frequencies that are unreproducible or attenuated in headphone speakers by emphasizing harmonics of the low bass frequencies. The high frequency enhancement can emphasize higher frequencies that may be less reproducible or poorly tuned for headphone speakers. In some implementations, the audio enhancement system provides a user interface that enables a user to control the amount (e.g., gains) of each enhancement applied to headphone input signals. The audio enhancement system may also be designed to provide one or more of these enhancements more effectively when headphones with good coupling to the ear are used.
Abstract: A method for transmitting an audio stream based on a focus of attention of a user within a multi-screen venue is presented. The method may include connecting to a mobile device associated with the user. The method may also include tracking a user face associated with the user by using at least one camera. The method may then include determining the focus of attention based on the tracked user face. The method may further include determining a video feed corresponding with the determined focus of attention. The method may also include transmitting the audio stream corresponding with the determined video feed to the mobile device.
Type:
Grant
Filed:
August 17, 2016
Date of Patent:
January 9, 2018
Assignee:
International Business Machines Corporation
Inventors:
Gregory J. Boss, Brent Hodges, John E. Moore, Jr., Sarbajit K. Rakshit
Abstract: A transmitting device comprises a binaural circuit (601) which provides a plurality of binaural rendering data sets, each binaural rendering data set comprising data representing parameters for a virtual position binaural rendering. Specifically, head related binaural transfer function data may be included in the data sets. A representation circuit (603) provides a representation indication for each of the data sets. The representation indication for a data set is indicative of the representation used by the data set. An output circuit (605) generates a bitstream comprising the data sets and the representation indications. The bitstream is received by a receiver (701) in a receiving device. A selector (703) selects a selected binaural rendering data set based on the representation indications and a capability of the apparatus, and an audio processor (707) processes the audio signal in response to data of the selected binaural rendering data set.
Type:
Grant
Filed:
December 10, 2013
Date of Patent:
January 2, 2018
Assignee:
KONINKLIJKE PHILIPS N.V.
Inventors:
Jeroen Gerardus Henricus Koppens, Arnoldus Werner Johannes Oomen, Erik Gosuinus Petrus Schuijers
Abstract: In one implementation, an apparatus includes a camera, a loudspeaker, a plurality of microphones, and a controller. The apparatus may be a computer, a mobile device such as a smart phone, or a dedicated videoconference device. The plurality of microphones are configured to produce sound data. A first number of the plurality of microphones are arranged on a first side of the loudspeaker, and a second number of the plurality of microphones are arranged on a second side of the loudspeaker. Different quantities of the microphones may be included on each side of the loudspeaker. The controller is configured to calculate a cross-correlation of the plurality of microphones based on the sound data and attenuate the cross-correlation based on a time-delay window and an attenuation factor. A participant location is determined based on the attenuated cross correlation functions.
Abstract: The present disclosure discloses a method and an apparatus for sound processing in a three-dimensional virtual scene. The method includes: acquiring, by a three-dimensional program engine, a sound processing request of a sound source point in a virtual scene; invoking a corresponding head-response transfer function (HRTF) according to a three-dimensional coordinate position relationship between the sound source point and a sound recipient in the virtual scene; modifying a parameter value of the HRTF according to the virtual scene where the sound source point is located; and performing filtering and delaying processing on a sound signal of the sound source point by using the modified HRTF.
Type:
Grant
Filed:
February 26, 2015
Date of Patent:
November 21, 2017
Assignee:
TENCENT TECHNOLOGY (SHENZHEN) COMPANY LIMITED
Abstract: 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.
Abstract: An audio signal processing apparatus includes: an obtaining unit which obtains a stereo signal including an R signal and an L signal; a control unit which generates a processed R signal and a processed L signal by performing (i) a first process of convolving pairs of right- and left-ear head related transfer functions into the R signal so that a sound image of the R signal is localized at each of two or more different positions at a right side of a listener; and (ii) a second process of convolving pairs of right- and left-ear head related transfer functions into the L signal so that a sound image of the L signal is localized at each of two or more different positions at a left side of the listener; and an output unit which outputs the processed R signal and the processed L signal.
Abstract: A method for perceptual enhancement of a received remote talker's voice, in the presence of local ambient noise, in an electronic voice communication system is provided. The method includes generating a pair of binaural voice signals from the remote talker's voice and manipulating the characteristics of the resulting pair of binaural signals. The two ears of the local listener are stimulated binaurally with the pair of binaural voice signals. The stimulating is performed adaptively as the ambient noise in the local listener's environment changes, thereby creating a perception of remote talker reacting actively to the local ambient noise, in a psycho-acoustically pleasing manner to the local listener.
Abstract: A first apparatus performs the following: determining, using microphone signals corresponding to a left microphone signal from a left microphone and a right microphone signal from a right microphone and using at least one further microphone signal, directional information of the left and right microphone signals corresponding to a location of a sound source; outputting a first signal corresponding to the left microphone signal; outputting a second signal corresponding to the right microphone signal; and outputting a third signal corresponding to the determined directional information.
Abstract: An apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information includes a parameter adjuster. The parameter adjuster is configured to receive one or more input parameters and to provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured to provide the one or more adjusted parameters in dependence on the one or more input parameters and the object-related parametric information, such that a distortion of the upmix signal representation caused by the use of non-optimal parameters is reduced at least for input parameters deviating from optimal parameters by more than a predetermined deviation.
Type:
Grant
Filed:
April 10, 2014
Date of Patent:
October 10, 2017
Assignees:
Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V., Dolby International AB, Friedrich-Alexander-Universitaet Erlangen-Nuernberg
Inventors:
Juergen Herre, Andreas Hoelzer, Leonid Terentiev, Thorsten Kastner, Cornelia Falch, Heiko Purnhagen, Jonas Engdegard, Falko Ridderbusch
Abstract: A low-quality rendition of a complex soundtrack is created, synchronized and combined with the soundtrack. The low-quality rendition may be monitored in mastering operations, for example, to control the removal, replacement or addition of aural content in the soundtrack without the need for expensive equipment that would otherwise be required to render the soundtrack.
Abstract: 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 partly based on the received at least one input signal, the simulated signal representing a simulation of at least one input signal reflecting from the ground or a 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 listener's ear level at a nominal listening position.