Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for collecting call data, feeding call data to applications, and providing advanced call features.

Abstract: An ultrasonic audio system includes adaptive equalization techniques configured to adjust the overall output of the ultrasonic audio system. This can be done by adjusting the equalization of the electronic audio signal representing the audio content being reproduced by the ultrasonic audio system. For example, in various embodiments, systems and methods can be implemented to increase the gain of higher frequency portions of the electronic audio signal while making a corresponding decrease in the amount of gain at the lower frequency portions of the same signal.

Abstract: A system for multiuser, geofixed acoustic simulations using a shared soundscape built and stored on a central server, via interfaces that also allow for live and competitive modification. A central server disseminates sound events from the soundscape to individual devices carried by users, which integrate geolocation information, head orientation information, and head-related transfer functions (HRTFs) to generate user-specific audio feeds. The audio feeds may be played through headsets so that users perceive the shared, geofixed sound events.

Abstract: Encoding of Higher Order Ambisonics (HOA) signals commonly results in high data rates. A method for low bit-rate encoding frames of an input HOA signal having coefficient sequences comprises computing (s110) a truncated HOA representation (CT(k)), determining (s111) active coefficient sequences (Ic,Act(k)), estimating (s16) candidate directions (MDIR(k)), dividing (s15) the input HOA signal into a plurality of frequency subbands (f1, . . . , fF), estimating (s161) for each of the frequency subbands a subset of candidate directions (MDIR(k)) as active directions (MDIR(k,f1), . . . , MDIR(k,fF)) and for each active direction a trajectory, computing (s17) for each frequency subband directional subband signals from the coefficient sequences of the frequency subband according to the active directions, calculating (s18) for each frequency subband a prediction matrix (A(k,f1), . . .

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

Abstract: The above sound reproduction device is applied to an acoustic space such as a passenger compartment, and controls the levels of the reproduced sounds at two evaluation points set at the seats in the passenger compartment, for example. Specifically, the sound reproduction device controls the phase of the sound signal inputted from external and supplies the sound signals to the pair of speakers. The phase control is performed for each of the plurality of listening positions such that a sum of reproduced sound levels at the two evaluation points becomes larger than the sum obtained in a case where the sound signal is reproduced only one of the pair of speakers.

Abstract: Disclosed is a system and method for generating a model of the geometric relationships between various audio sources recorded by a multi-camera system. The spatial audio scene module associates source signals, extracted from recorded audio, of audio sources to visual objects identified in videos recorded by one or more cameras. This association may be based on estimated positions of the audio sources based on relative signal gains and delays of the source signal received at each microphone. The estimated positions of audio sources are tracked indirectly by tracking the associated visual objects with computer vision. A virtual microphone module may receive a position for a virtual microphone and synthesize a signal corresponding to the virtual microphone position based on the estimated positions of the audio sources.

Abstract: Provided is a method and devices for sensing a flip cover including determining an angle of the flip cover. A Hall sensor may be included in a mobile terminal (or display terminal) and may include at least one Hall element for collecting a magnetic field intensity generated from a magnet of the flip cover.

Abstract: In general, techniques are described for determining renderers used for rendering spherical harmonic coefficients to generate one or more loudspeaker signals. A device comprising one or more processors may perform the techniques. The one or more processors may be configured to determine a local speaker geometry of one or more speakers used for playback of spherical harmonic coefficients representative of a sound field, and configure the device to operate based on the local speaker geometry.

Abstract: An example game device corrects a sound of a sound source located behind a virtual camera by applying a frequency filter to the sound. Specifically, an angle between an imaging direction of the virtual camera and a direction from the virtual camera toward the sound source is calculated. If the absolute value of the angle is 90 degrees or more, a filter value is calculated based on the angle. The game device also corrects the filter value based on a distance between the virtual camera and the sound source. The game device applies a low-pass filter corresponding to the filter value to the sound of the sound source.

Abstract: An apparatus for generating an output signal having at least two output channels from an input signal having at least two input channels, has an ambient/direct decomposer, an ambient modification unit and a combination unit. The ambient/direct decomposer is adapted to decompose at least two input channels of the input signal such that each one of the at least two input channels is decomposed into a signal of a first signal group and into a signal of a second signal group. The ambient modification unit is adapted to modify a signal of the ambient signal group or a signal derived from a signal of the ambient signal group to obtain a modified signal as a first output channel. The combination unit is adapted to combine a signal of the ambient signal group or a signal derived from a signal of the ambient signal group and a signal of the direct signal group or a signal derived from a signal of the direct signal group as a second output channel.

Abstract: Embodiments of systems and methods are described for providing backwards compatibility for legacy devices that are unable to natively render non-channel based audio objects. These systems and methods can also be beneficially used to produce a reduced set of audio objects for compatible object-based decoders with low computing resources.

Abstract: Improved methods and devices for processing low-frequency audio data are provided. A bass extraction process may involve applying low-pass filters to received audio object signals, to produce extracted low-frequency audio signals. The bass extraction process may be performed prior to a process of rendering audio objects into speaker feed signals. A bass management process may involve routing the extracted low-frequency audio signals to the one or more speakers capable of reproducing low-frequency audio signals.

Abstract: A mobile terminal and controlling method thereof are disclosed, by which a content sharing between a mobile terminal and a call counterpart is facilitated in the course of making a phone call in-between. The present invention includes a wireless communication unit configured to perform a communication with an external terminal device while the external terminal is currently in a phone call with a call counterpart, a touchscreen configured to display a content list of contents to share with the call counterpart, and a controller to control the touchscreen, wherein in response to receiving a user input to request a sharing of at least one content in the content list, the controller to control a request, to the external terminal device, to share the sharing requested content with the call counterpart.

Abstract: An audio headset may receive a plurality of audio signals corresponding to plurality of surround sound channels. The headset may determine, via its audio processing circuitry, context and content of the audio signals. The audio processing circuitry may process the audio signals to generate a pair of stereo signals carrying one or more virtual surround channels, wherein the processing comprises automatically controlling, based on the context and the content of the audio signals, a simulated acoustic environment of the virtual surround channels.

Abstract: A method for generating an auditory environment for a user may include receiving a signal representing an ambient auditory environment of the user, processing the signal using a microprocessor to identify at least one of a plurality of types of sounds in the ambient auditory environment, receiving user preferences corresponding to each of the plurality of types of sounds, modifying the signal for each type of sound in the ambient auditory environment based on the corresponding user preference, and outputting the modified signal to at least one speaker to generate the auditory environment for the user. A system may include a wearable device having speakers, microphones, and various other sensors to detect a noise context. A microprocessor processes ambient sounds and generates modified audio signals using attenuation, amplification, cancellation, and/or equalization based on user preferences associated with particular types of sounds.

Abstract: In example embodiments, a system and method for presenting a combined audio stream from a single presentation system to a plurality of users is provided. The method comprises receiving an audio filtering signal, the audio filtering signal corresponding to unwanted audio data to be filtered from an ambient audio signal that includes a combined audio stream output from an audio output device of the single presentation system. A device filtering signal is generated based on the audio filtering signal. The device filtering signal is output by an audio device of a user (e.g., headphone). The device filtering signal is played through the audio device to filter out the unwanted audio data while allowing other audio data in the ambient audio signal to be heard by the user.

Abstract: Methods, systems, and apparatus are generally described for providing an audio interface. In some examples, first voice data of a first narrator and a second voice data of a second narrator are received and the second voice data is transformed by a voice transformation function. At least a part of a first text data is converted into a first synthesized voice data based, at least in part, on the first voice data and at least a part of a second text data is converted into a second synthesized voice data based, at least in part, on the transformed second voice data by applying a voice transformation function which maximizes a feature difference between the first voice data and the transformed second voice data. The first synthesized voice data and the second synthesized voice data are provided in parallel on a temporal axis via the voice interface system.

Abstract: Provided is an audio signal processing device including a signal processing section that changes, at a time of generating and outputting 2-channel audio signals to be subjected to sound reproduction by two electroacoustic transducing means located at positions in the vicinities of both ears of a listener, from audio signals of a plurality of and more than two channels, virtual sound image localization positions on a circle around the listener, across the virtual sound image localization positions, the virtual sound image localization position that is supposed for each of the plurality of channels of audio signals provided on the circle.

Abstract: A method comprises providing at least one processing unit comprising a decomposing section and a playback section; receiving, at the decomposing section, audio data generated via an array of microphones, the audio data representing an acoustic scene; decomposing the audio data into a plurality of signals representing components of the acoustic scene arriving from a plurality of directions, using the decomposing section; and rendering the audio components for a listener based on the plurality of directions of the audio components, using the playback section.

Abstract: A voice controlled assistant has two speakers to produce high quality stereo sound. The voice controlled assistant generates a first frequency portion and a second frequency portion used to produce stereo sound. The frequency portion is filtered to remove a range of frequencies, such as the low frequency range. The second frequency portion, together with the frequency portion removed from the first frequency portion, are output as sound.

Abstract: An apparatus comprising: a perception sorter configured to perceptually order at least two object orientated audio signal channels; and a selective channel processor configured to process at least one of the at least two object orientated audio signal channels based on the order of the at least two object orientated audio signal channels.

Abstract: A method and apparatus for providing improved intelligibility of contemporaneously perceived audio signals. Differentiation cues are added to monaural audio signals to allow a listener to more effectively comprehend information contained in one or more of the signals. In a specific embodiment, a listener wearing stereo headphones listens to simultaneous monaural radio broadcasts from different stations. A differentiation cue is added to at least one of the audio signals from the radio reception to allow the listener to more effectively focus on and differentiate between the broadcasts.

Abstract: The disclosure includes a system and method for sonically customizing an audio reproduction device. The system includes a processor and a memory storing instructions that when executed cause the system to determine an application environment associated with an audio reproduction device associated with a user; determine one or more sound profiles based on the application environment; provide the one or more sound profiles to the user; receive a selection of a first sound profile from the one or more sound profiles; and generate tuning data based on the first sound profile, the tuning data configured to sonically customize the audio reproduction device.

Abstract: A method improves computer performance to provide binaural sound in a telephone call. The method includes providing the telephone call with a voice of a calling party and a coordinate location that describes where a voice of the calling party should localize as binaural sound to the called party. The voice of the calling party is convolved to localize at the coordinate location.

Abstract: An audio system comprises at least one processor configured to receive multi-channel audio data and comprising a plurality of output channels. Control instructions are established based on preset positions of a plurality of speakers connectable to the output channels and further based on virtual positions of the channels of the multi-channel audio data. The channels of the multi-channel audio data are routed to the output channels based on the control instructions.

Abstract: Systems and methods facilitating removal of content from audio files are described. A method includes identifying a sound recording in a first audio file, identifying a reference file having at least a defined level of similarity to the sound recording, and processing the first audio file to remove the sound recording and generate a second audio file. In some embodiments, winner-take-all coding and Hough transforms are employed for determining alignment and rate adjustment of the reference file in the first audio file. After alignment, the reference file is filtered in the frequency domain to increase similarity between the reference file and the sound recording. The frequency domain representation (FR) of the filtered version is subtracted from the FR first audio and the result converted to a time representation of the second audio file. In some embodiments, spectral subtraction is also performed to generate a further improved second audio file.

Abstract: The present disclosure relates to a positioning method and apparatus in a three-dimensional space of reverberation. The method includes the following steps: constructing a three-dimensional space frame surrounding a point sound source and a sound receiver, and acquiring barycentric coordinates of the three-dimensional space frame; establishing an X-Y coordinate system by using the barycentric coordinates as the origin; determining a vector from the sound receiver to the barycentric coordinates and a facing angle between the sound receiver and the barycentric coordinates according to the established X-Y coordinate system; and obtaining reverberation intensity through calculation according to the vector from the sound receiver to the barycentric coordinates and the facing angle between the sound receiver and the barycentric coordinates, and using the reverberation intensity as a control parameter.

Abstract: A device comprising one or more processors is configured to determine a plurality of segments for each of a plurality of binaural room impulse response filters, wherein each of the plurality of binaural room impulse response filters comprises a residual room response segment and at least one direction-dependent segment for which a filter response depends on a location within a sound field; transform each of at least one direction-dependent segment of the plurality of binaural room impulse response filters to a domain corresponding to a domain of a plurality of hierarchical elements to generate a plurality of transformed binaural room impulse response filters, wherein the plurality of hierarchical elements describe a sound field; and perform a fast convolution of the plurality of transformed binaural room impulse response filters and the plurality of hierarchical elements to render the sound field.

Abstract: A device for calculating loudspeaker signals using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal to a spectral domain to obtain a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum for a combination consisting of a loudspeaker and an audio signal based on a delay value, a filter stage for filtering the specific short-term spectrum by using a filter, so that a filtered short-term spectrum is obtained for each audio signal and loudspeaker combination, a summing stage for summing up the filtered short-term spectra for a loudspeaker to obtain summed-up short-term spectra, and a backtransform stage for backtransforming summed-up short-term spectra for the loudspeakers to a time domain to obtain the loudspeaker signals.

Abstract: In some embodiments, portable speakers can be small and lightweight and can communicate with one or more audio device over wired or wireless connections. In some embodiments, portable speakers achieve reduced complexity as compared to typical high fidelity systems (e.g., by including a reduced number of speaker drivers and amplifiers), while still maintaining high fidelity stereo audio playback, thereby achieving both portability and high quality audio capability. For instance, certain implementations of the speaker include two primary speakers disposed on opposing faces of the speaker enclosure (e.g., full or mid-range drivers) and two tweeters, also disposed on opposing faces. Primary speakers can be disposed on respective ends of the housing and each can output a different stereo channel. Each tweeter can be positioned on different face of the housing. The speaker system according to some embodiments generates a mono high frequency signal to drive the tweeters.

Abstract: An electronic device includes processing circuitry configured to produce an audio sound from a virtual sound source position. The circuitry also acquires from a user-input a sound position information of a perceived sound source position, and then controls an enable or disable of a function of the electronic device based on a relationship between the virtual sound source position and the perceived sound source position.

Abstract: Provided is an audio processing device including a narration canceling section configured to generate a narration canceling signal by removing a narration component from an input signal, and a reverberation adding section configured to add a reverberation effect to the narration canceling signal.

Abstract: The invention provides a method and device of extracting a sound source acoustic image body in 3D space. The method includes: determining a spatial position of a sound source acoustic image and determining a speaker beside the spatial position where the sound source acoustic image is located according to the determined spatial position (?, ?, ?) of the sound source acoustic image; calculating a correlation of signals of all sound tracks of the selected speaker in the horizontal direction and the vertical direction, and obtaining and storing a parameter set {ICH, ICv, Min{ICH, ICv}} of a acoustic image body, wherein the Min{ICH, ICv} is a smaller value between ICH and ICV. The expression parameters of the acoustic image body obtained in the present invention are used for providing technical support for accurately restoring the size of the sound source acoustic image in a 3D audio live system, which solves the technical problem that the restored acoustic image in a 3D audio is excessively narrow at present.

Abstract: A system calibrates one or more head-related transfer functions (HRTFs) for a user. An indicator is presented on a head-mounted display, where the indicator prompts the user to turn the user's head to view the indicator and effectively change the user's head orientation. The head orientation corresponds to positions of both ears, thus a position of the indicator prompting to change the user's head orientation is associated with corresponding positions of both ears. Responsive to the indicator being viewed by the user, a sound source at a fixed position transmits a test sound and the test sound is received at microphones coupled to the user's ears. By analyzing the test sound received at the microphones, a unique HRTF associated with a relative position between the sound source and each ear can be obtained.

Abstract: An embodiment in accordance with the present invention provides a system and method for binaural rendering of complex acoustic scenes. The system includes a computing device configured to process a sound recording of the acoustic scene to produce a binaurally rendered scene for the listener. The system also includes a position sensor configured to collect motion and position data for a head of the user and also configured to transmit said motion and position data to the computing device. A sound delivery device is configured to receive the binaurally rendered acoustic scene from the computing device and to transmit the acoustic scene to the ears of the listener. In the system, the computing device is further configured to utilize the motion and position data from the inertial motion sensor to process the sound recording of the acoustic scene with respect to the motion and position of the user's head.

Abstract: Example disclosed media source detection methods include determining first trending coefficients for a first audio signal obtained from a first media source in communication with a media device, wherein the first trending coefficients correspond to a set of frequency bands. Example disclosed media source detection methods also include determining second trending coefficients for a second audio signal obtained from monitoring media presented by the media device, wherein the second trending coefficients correspond to the set of frequency bands.

Abstract: Method and devices for providing surround audio signals are provided. Surround audio signals are received and are binaurally filtered by at least one filter unit. In some embodiments, the input surround audio signals are also processed by at least one equalizing unit. In those embodiments, the binaurally filtered signals and the equalized signals are combined to form output signals.

Abstract: The invention relates to a method for processing a multichannel sound in a multichannel sound system, wherein the input signals L and R are decoded, preferably as stereo signals. The aim of the invention is to develop the method such that a further improvement of the spatial reproduction of the input signals L and R is achieved on the basis of an extraction of direction components. According to the invention, this is achieved in that the signals R and L are decoded at least into two signals of the form nL-mR, in which n, m=1, 2, 3, 4.

Abstract: A reproducing device includes a reproducing unit configured to reproduce a contents sound, an output unit configured to output the contents sound which has been reproduced using the reproducing unit to a headphone, a talk detection unit configured to detect that a headphone fitter who fits the headphone talks with a person and an image normal position control unit configured to move the normal position of the image of the contents sound which has been reproduced using the reproducing unit to an arbitrary position when the talk detecting unit has detected that the headphone fitter has started talking with the person.

Abstract: 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.

Abstract: An audio-signal processing device includes a decoding unit that decodes a compressed audio stream to obtain audio signals for a predetermined number of channels; a signal processing unit that generates 2-channel audio signals including left-channel audio signals and right-channel audio signals, on the basis of the predetermined-number-of-channels audio signals; and a coefficient setting unit that sets filter coefficients corresponding to the impulse responses for the digital filters, on the basis of format information of the compressed audio stream. The signal processing unit uses digital filters to convolve impulse responses for paths from sound-source positions of the channels to the left and right ears of a listener with the corresponding predetermined-number-of-channels audio signals and adds corresponding results of the convolutions for the channels to generate the left-channel audio signals and the right-channel audio signals.

Abstract: An information processing apparatus includes: position detection means for detecting a position of a client unit held by a user on the basis of a signal outputted from the client unit; conversion means for variably setting a parameter value used to convert at least one of a sound signal and a video signal on the basis of the position of the client unit detected by the position detection means and converting the signal using the parameter value; and output means for outputting the signal after conversion by the conversion means.

Abstract: The present invention relates to an audio signal processing apparatus and an audio signal processing method which perform binaural rendering.

Abstract: A method captures audio impulse responses of a person wearing electronic microphones in a left ear and a right ear. A smartphone generates a sound while it is away from but proximate to the person. The microphones capture the audio impulse responses that are used to convolve sound so it localizes away from but proximate to the person.

Abstract: The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.

Abstract: A method and apparatus are disclosed to determine individualized head-related transfer function (HRTF) parameters for a user. The technique can include determining HRTF data of a user by using transform data of the user, where the transform data is indicative of a difference, as perceived by the user, between a sound of a direct utterance by the user and a sound of an indirect utterance by the user. The technique may further involve producing an audio effect tailored for the user by processing audio data based on the HRTF data of the user.

Abstract: A signal processing method and system are provided for delivering spatialized sound using highly optimized inverse filters to deliver narrow localized beams of sound from the included speaker array. The inventive method can be used to provide private listening areas in a public space and provide spatialization of source material for individual users to create a virtual 3D audio effect. In a binaural mode, a speaker array provides two targeted beams aimed towards the primary user's ears—one discrete beam for the left ear and one discrete beam for the right ear.

Abstract: The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.

Abstract: The present technique relates to a sound processing apparatus, a sound processing method, and a program which are configured to obtain higher quality sound. In order to obtain an objective characteristic, the sound processing apparatus previously holds a characteristic of the sound processing apparatus, generates an inverse characteristic filter of the characteristic of the sound processing apparatus, and generates a correction filter from the obtained filter and a filter having the objective characteristic. The sound processing apparatus uses the generated correction filter to subject a sound signal to be reproduced to filter processing, and reproduces sound based on the obtained sound signal. Therefore, a characteristic to be originally obtained can be achieved. The present technique can be applied to the sound processing apparatus.