Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: Certain embodiments provide methods and systems for managing a sound profile. An example playback device includes a network interface and a non-transitory computer readable storage medium having stored therein instructions executable by the processor. When executed by the processor, the instructions are to configure the playback device to receive, via the network interface over a local area network (LAN) from a controller device, an instruction. The example playback device is to obtain, based on the instruction, via the network interface from a location outside of the LAN, data comprising a sound profile. The example playback device is to update one or more parameters at the playback device based on the sound profile. The example playback device is to play back an audio signal according to the sound profile.

Abstract: An electronic apparatus and method for configuration of an audio reproduction system is provided. The electronic apparatus receives image of a listening environment and applies a machine learning model on the image to identify a plurality of objects, including a display device and a plurality of audio devices. The electronic apparatus determines contour information of each identified object. The electronic apparatus retrieves real-dimension information of each identified object. Based on the determined contour information and the retrieved real-dimension information, the electronic apparatus determines first distance information between a listening position and each identified object. The electronic apparatus receives an audio signal from each audio device and determines a second distance between each audio device and the listening position based on the received audio signal.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization based on variant selection. An example apparatus to generate equalization adjustments for an audio signal includes an input feature set generator to generate an equalization input feature set by generating a data structure corresponding to a user input, the data structure including a number of entries identifying a selected variant of music and including the data structure in the equalization input feature set, and a model executor to, in response to obtaining the equalization input feature set, adjust at least one weight of a neural network model to generate the equalization adjustments for the audio signal.

Abstract: A vehicle includes a first voice output section, a second voice output section, a third voice output section, and a control section which are arranged side by side in a width direction of the vehicle. The control section controls volume of the voice output from the first, second and third voice output sections. If sitting on a rider seat and a pillion passenger seat is detected, the control section controls the first, second and third voice output sections to make the volume of the voice output from the first and third voice output sections larger than the volume of the voice output from the second voice output section. The first and third voice output sections are located at both ends in the width direction of the vehicle and the second voice output section is located between the first and third voice output sections.

Abstract: A method for adjusting sound playback of a portable device for constancy notwithstanding different environments outputs from the portable device detectable audio signals inaudible to user and the device receives reflected audio before the portable device is actually commanded to play an audio file. A list of volume weightings for reflected audio is calculated. Before commencing playback of the audio file, the portable device obtains reference volume weightings from a list according to the current volume setting, and calculates adjustment coefficients for different frequency bands based on weightings of the reference volume list and of the reflected audio list. The audio signals of the audio file are output after adjustment. A portable device is also disclosed.

Abstract: The invention discloses a prediction method for porous material of electroacoustic devices and prediction system thereof. The method comprises the following steps. The step (A) is to obtain at least one acoustic parameter of a porous material from an electroacoustic device, and the at least one acoustic parameter comprises a flow resistance value, a specific flow resistance value and a flow resistance ratio. The step (B) is to calculate an actual resistance value of the porous material based on the at least one acoustic parameter. Thereafter, the step (C) establishes an equivalent circuit model corresponding to the electroacoustic device based on the structure configuration and material parameters of the electroacoustic device. At last, step (D) introduces the actual impedance value of the porous material into the equivalent circuit model, and calculates the frequency response curve and impedance curve of the electroacoustic device affected by the porous material.

Abstract: A mixing device of a first signal and a second signal on a time-frequency plane, includes a control signal generation unit to generate a control signal indicating whether to perform prioritized mixing that includes amplification of the first signal and attenuation of the second signal; and a gain derivation unit to derive a first gain for amplifying the first signal and a second gain for attenuating the second signal based on the control signal. The control signal takes at least a first value and a second value different from the first value, wherein the first value is not continuous beyond a predetermined bandwidth on a frequency axis. The mixing device applies the prioritized mixing to the first and second signals when the control signal indicates the first value, and applies simple addition to the first and second signals when the control signal indicates the second value.

Abstract: The present application relates to a loudness adjustment method and apparatus, and an electronic device and a storage medium, mainly relating to the technical field of multimedia. The method comprises: converting a sound signal of a multimedia resource into a first frequency domain signal; acquiring a second frequency domain signal based on the first frequency domain signal and frequency response information of a current electronic device, wherein the second frequency domain signal is used for reflecting the loudness of the first frequency domain signal when same is played on the current electronic device; and adjusting the loudness of the sound signal based on the second frequency domain signal and a target loudness, so as to obtain a target sound signal of the multimedia resource.

Abstract: A microphone capsule assembly includes a first microphone capsule oriented to face a first direction that is parallel to a first plane, a second microphone capsule oriented to face in a direction opposite to the first direction, where a front face of the second microphone capsule is spaced a distance from a front face of the first microphone capsule in the first direction, a third microphone capsule oriented to face in a second direction that is at a first acute angle to the first direction when measured parallel to the first plane, and a fourth microphone capsule oriented to face in a third direction that is at a second acute angle to the first direction when measured parallel to the first plane, where the third and fourth microphone capsules are spaced a distance from the front face of the first microphone capsule in the first direction.

Abstract: A hearing aid system is provided that facilitates adjustment of signal processing parameters ? of the hearing aid system with minimum user intervention, wherein the hearing aid system is capable of calculating signal processing parameters ? for evaluation of the user when the user has entered an input, e.g. using a smartwatch, to this effect. The evaluation takes place for a certain time period and in the event that the user has entered a consent input indicating that he or she is pleased with the set ? of signal processing parameters under evaluation, the hearing aid system continues processing with those signal processing parameters; and if the user is not pleased with the signal processing parameters ? under evaluation, the hearing aid system calculates another set {circumflex over (?)} of signal processing parameters for evaluation of the user.

Abstract: A mixing apparatus comprises a weighting circuit configured to, within a particular control target band, perform weighting amplitude spectra of a priority audio signal and a non-priority audio signal taking frequency masking and time masking into account, an amplitude changer configured to relatively amplify an amplitude spectrum of the priority audio signal within the control target band based on the amplitude spectrum to which the weighting has been applied, a phase changer configured to make a phase spectrum of the non-priority audio signal approach a phase spectrum of the priority audio signal, within the control target band, based on the amplitude spectrum to which the weighting has been applied, and a mixer configured to mix the priority audio signal and the non-priority audio signal after the amplitude spectrum and the phase spectrum are changed.

Abstract: Determination of an acoustic filter for incorporating local effects of room modes within a target area is presented herein. A model of the target area is determined based in part on a three-dimensional virtual representation of the target area. In some embodiments, the model is selected from a group of candidate models. Room modes of the target area are determined based on a shape and/or dimensions of the model. The room mode parameters are determined based on at least one of the room modes and the position of a user within the target area. The room mode parameters describe an acoustic filter that as applied to audio content, simulates acoustic distortion at the position of the user and at frequencies associated with the at least one room mode. The acoustic filter is generated at a headset based on the room mode parameter and is used to present audio content.

Abstract: An example computing device is configured to perform functions including receiving calibration data corresponding respectively to a plurality of playback devices, where each playback device in the plurality of playback devices is located in a respective playback environment other than a first playback environment. The functions also include receiving playback device characteristic data respectively indicating at least one playback device characteristic for each playback device of the plurality of playback devices. The functions also include, based on at least the calibration data and the playback device characteristic data, determining updated playback device configuration information and transmitting data indicating the updated playback device configuration information to a given playback device located in the first environment.

Abstract: One embodiment provides a method of personalized headphone equalization system for a headphone. The method comprises obtaining a measurement of sound pressure level at a microphone mounted in a near field of a headphone driver of the headphone and inside a cavity formed by the headphone and a user's ear. The method further comprises providing personalized equalization (EQ) of output reproduced via a headphone driver by performing EQ correction of the output based on the measurement of sound pressure level and a pre-determined target frequency response for the headphone, resulting in an equalized output that is adapted to individual characteristics of the user's ear.

Abstract: Systems, methods, apparatus, and articles of manufacture to control audio playback devices are disclosed. An example first playback device includes a speaker driver, a processor, and a computer readable medium including a set of instructions that, when executed by the processor, cause the first playback device to implement a method. The example method includes receiving, from a first audio information source, first audio information. The example method includes playing back the first audio information. The example method includes receiving, from a second audio information source, (i) a first message, and (ii) second audio information.

Abstract: Reduced order modeling is used in the present systems and methods for monitoring the health of a system and is particularly useful in monitoring the health and estimating the remaining useful life of blow out preventer (BOP) annular used in an oil or gas well. A reduced order model is a linear low order characterization of the system that describes the annular behavior with a small number of parameters that can be interpreted physically. The reduced order model is then identified and adapted in real-time in order to detect the change in the system behavior and characteristics. This adaptation helps in drawing conclusions on diagnostics and prognostics of the physical system.

Abstract: Example techniques facilitate calibration of a playback device. An example implementation involves a computing device capturing, via a microphone, data representing multiple iterations of a calibration sound as played by a playback device. The computing device identifies multiple sections within the captured data. Two or more sections represent respective iterations of the calibration sound as played by the playback device. Based on the multiple identified sections, the computing device determines a frequency response of the playback device, the frequency response of the playback device representing audio output by the playback device and acoustic characteristics of an environment around the playback device. Based on the frequency response of the playback device and a target frequency response, the computing device determines one or more parameters of an audio processing algorithm and sends, to the playback device, the one or more parameters of the audio processing algorithm.

Abstract: A determining method that makes it possible to determine a phase filter for a system that generates vibrations and that includes a first transducer for converting a first electrical signal and a second transducer for converting a second electrical signal. The method includes: performing, for at least one perception position, of a plurality of spectral measurements of a characteristic parameter of the vibrations generated in this perception position as a function of the frequency, each spectral measurement being done for a respective phase shift value between the first electrical signal and the second electrical signal; and determining a phase filter from spectral measurements done, by selecting, for each frequency, a phase shift value from among the phase shift values used to perform the spectral measurements.

Abstract: A variable-resolution graphic equalizer providing an improved interface for controlling gain values across the entire audio spectrum using many narrow-band filters (e.g., 120). It allows user selection of a frequency range for graphic equalization and automatically maps a reduced and fixed number of sliders to the selected range based on the number of filter bands falling within the selected range. In an audio processing system, specific user interface regions are highlighted to display selected frequency ranges and corresponding selected sliders to allow for rapid and precise equalization of the full audio spectrum using the many narrow-band filters.

Abstract: Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, while a playback device is a part of a first zone group that includes the playback device and at least one first playback device, the playback device applies a first audio setting. The embodiment also includes the playback device joining a second zone group that includes the playback device and at least one second playback device. The embodiment further includes the playback device applying a second audio setting based on an audio content profile corresponding to the second zone group.

Abstract: Acoustic radiation control method and system are provided. The acoustic radiation control method includes: configuring a speaker array; obtaining transfer functions of speakers in the speaker array based on configuration of the speaker array and directivity of the speakers; obtaining, based on the transfer functions of the speakers, source strength of the speakers which enables acoustic radiation of the speaker array in a first zone greater than acoustic radiation of the speaker array in a second zone; and applying the source strength of the speakers to the speaker array. By the method, acoustic radiation may be controlled more accurately, a sidelobe level may be constrained more effectively, and the number of speakers in the speaker array may be reduced.

Abstract: An apparatus for processing an audio signal for reproduction by a sound transducer includes an equalization parameter determinator for determining a set of equalization parameters and an equalizer configured to equalize an input audio signal, to obtain an equalized audio signal. Different concepts for the determination of the set of equalization parameters include an image recognition, an evaluation of an identification signal which is provided by the sound transducer via an audio connection, and a measurement of the impedance of the sound transducer over frequency. Also, an upload functionality and a download functionality are provided.

Abstract: Disclosed in the present specification are a multimedia apparatus for servicing an optimized sound depending on the surrounding environment and a method for processing an audio signal thereof. The method for processing an audio signal of the multimedia apparatus, according to the present invention, comprises the steps of: receiving an external test sound; analyzing the frequency properties of the received test sound; calculating a compensation value of the test sound according to the analyzed frequency properties; compensating an audio signal to be output, according to the calculated compensation value; and outputting the compensated audio signal.

Abstract: Example techniques involve suggesting an application of a streaming audio service via an application of a media playback system. An example implementation involves a mobile device displaying, via a control application of a media playback system, a library control comprising indicators representing respective media items of a particular streaming audio service. The mobile device receives, via the library control, input data indicating a selection of an indicator representing a particular media item. The mobile device sends instructions to cause a playback device to play back the particular media item. While the playback device is playing back the particular media item, the mobile device displays, via the control application, a control representing an application of the particular streaming audio service. The mobile device receives input data indicating a selection of the control representing the application of the particular streaming audio service and causes the mobile device to download the application.

Abstract: A method for generating an individualized audio output response for a headset based on images of a user's ear. An image of a portion of a user's head including at least the user's ear is received, the user in the image wearing a headset including a plurality of visual markers. One or more features describing the user's ear are identified based at least in part on a position of one of the plurality of visual markers relative to the user's ear that are used to disambiguate orientation and/or scale of the features in the image. The one or more features are input to a model, and the model is configured to determine an audio output response of the user based on the extracted one or more features. An individualized audio output response is generated for the user based on the audio output response, the individualized audio output response configured to adjust one or more acoustic parameters of audio content provided to the user by the headset.

Abstract: Systems and methods herein utilize psychoacoustic principles to process audio signals to achieve a desired result. In some embodiments, an exemplary process utilizes a combination of re-harmonization and auditory processing techniques in which a plurality of filtered audio cues are sampled, created and down-mixed into a final music output, which may be designed to trigger specific effects on the function of the body and the brain. In some embodiments, enhanced audio material may be altered during the stream or fixed for a desired result at the beginning of the stream. In some embodiments, a signal enhancement process may be embedded into silicon or a media player, decoder or device with software. In other embodiments, audio or video material may be encoded or finalized into the media file.

Abstract: A portable speaker, such as a subwoofer, that is configured to pair via Bluetooth from any device capable of streaming Bluetooth audio and act as the “sink.” The portable speaker is also configured to pair via Bluetooth to other audio equipment capable of streaming Bluetooth audio and act as the “source.” The portable speaker is also preferably configured to synchronize the audio signal going to the portable speaker and the other unit playing music, such that the user will not hear delays between the two units playing the audio stream. Preferably, the portable speaker is configured to be portable in which the size and weight is small enough for a person to carry, and is portable in which it operates with rechargeable batteries inside the unit, thus not needing any other power source such as AC outlet, or automobile power system. Another embodiment omits the speaker and instead provides a line out for connection to an external speaker, such as a subwoofer.

Abstract: Methods and systems for generating a speaker feed for driving a sealed speaker, including: generating feedback indicative of excursion of the speaker; in response to the feedback, equalizing an input signal to generate the speaker feed, such that the speaker feed is equalized for driving the speaker with: a desired frequency response above the speaker's resonance frequency; and an at least substantially flat frequency response in a variable sub-resonance equalization band without exceeding the speaker's excursion limit, where the sub-resonance equalization band extends from the resonance frequency down to a variable cutoff frequency, and the cutoff frequency is determined in response to the feedback.

Abstract: The present invention discloses a sound signal processing method, a terminal, and a headset, to prevent a sound in an external environment from causing unnecessary disturbance to a user, thereby improving user experience. The terminal includes: a microphone, configured to collect an ambient sound signal; and a processor, configured to: obtain the sound signal collected by the microphone, and process, based on user status information, the ambient sound signal collected by the microphone, where the user status information includes a geographical location of a user who uses the terminal or a motion status of the user.

Abstract: A propagation delay identification system for active noise cancelation for a vehicle audio system may include at least one sensor configured to transmit sensor data indicative of acceleration data, hammer data and microphone data, at least one output sensor configured to transmit an impulse response, and a processor. The processor may be programmed to receive the sensor data and pulse function, identify a propagation path delay between the microphone data and a cross-correlation between the acceleration data and hammer data, and generate a source pulse function. The processor further identify a secondary path delay based on an arrival time of peak energy of a convolution of the impulse response and filtered pulse function, and apply a binary limiter to a reference signal in response to the secondary path delay exceeding the propagation path delay to reduce boosting of an audio signal based on coherent reference and error signals.

Abstract: Example techniques relate to calibration interfaces that facilitate calibration of a playback device. An example implementation may involve displaying, on a graphical display, a sequence of control interfaces to guide calibration of a playback device during a calibration sequence comprising (i) a spatial calibration component and (ii) a spectral calibration component. Displaying the sequence of control interfaces includes displaying one or more spatial calibration interfaces comprising a guide to perform the spatial calibration component of the calibration sequence. The spatial calibration component involves calibration of the playback device for a particular location within an environment. Displaying the sequence of control interfaces also includes displaying one or more spectral calibration interfaces comprising a guide to perform the spectral calibration component of the calibration sequence. The spectral calibration component involves calibration of the playback device for the environment.

Abstract: An example playback device is configured to receive a first stream of audio comprising source audio content to be played back by the playback device and record, via one or more microphones of the playback device, an audio signal output by the playback device based on the playback device playing the source audio content. The playback device is also configured to determine a transfer function between a frequency-domain representation of the first stream of audio and a frequency-domain representation of the recorded audio signal, and then determine an estimated frequency response of the playback device based on a difference between (i) the transfer function and (ii) a self-response of the playback device, where the self-response of the playback device is stored in a memory of the playback device. Based on the estimated frequency response, the playback device is configured to determine an acoustic calibration adjustment and implement the acoustic calibration adjustment.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal to reproduce high quality sounds on various materials. In various embodiments, a method comprises filtering the signal with a low shelf filter and/or high shelf filter, passing the signal through a first compressor that, filtering the signal again with a low shelf filter and/or high shelf filter, processing the signal with a graphic equalizer based on a selected material profile, passing the signal through a second compressor, and outputting the signal to a transducer.

Abstract: Playing back of a piece of multimedia sound content by means of the vibration of any radiating structure, produced by contact with a playback device comprising: a mechanical actuator applying vibratory excitation to the structure, from a source signal, corresponding to a multimedia sound signal, a sensor sensing a signal from the vibration of the radiating structure, in order to thus: supply the actuator with a source signal, of known frequency components, receive, from the sensor, a vibration signal from the radiating structure operated by the actuator, in order to obtain a frequency response from the radiating structure, correct the source signal supplied to the actuator, by applying a filter correcting the spectral response of the radiating structure such that this response, once corrected, has a flat spectral profile, with amplitude deviations at most of 5 dB, and supply the actuator with the source signal corrected in this way.

Abstract: The system and method includes positioning a mobile device with a built-in loudspeaker at a first location in a listening environment and at least one microphone at at least one second location in the listening environment; emitting test audio content from the loudspeaker of the mobile device at the first position in the listening environment; receiving the test audio content emitted by the loudspeaker using the at least one microphone at the at least one second location in the listening environment; and, based at least in part on the received test audio content, determining one or more adjustments to be applied to desired audio content before playback by at least one earphone; wherein the first location and the second location are distant from each other so that the at least one microphone is within the near-field of the loudspeaker.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization based on variant selection. An example apparatus to generate equalization adjustments for an audio signal based on user input includes a user interface to prompt a user for the user input corresponding to a selected variant of music, and an input feature set generator to generate an equalization input feature set, the equalization input feature set to be used by a model executor to adjust at least one weight of a neural network model to generate example equalization adjustments for the audio signal based on the user input, the equalization input feature set generated by: generating a data structure corresponding to the user input, the data structure including a number of entries identifying the selected variant of music, and including the data structure in the equalization input feature set.

Abstract: When the volume is adjusted in a multi-speaker system, it is desirable that one speaker does not change volume disproportionately with respect to another speaker. A method is presented for adjusting a volume level of one or more speakers. Each speaker can have a non-standardized relationship between logical volume level that is input to the speaker and sound pressure level that is produced by the speaker. A selected volume level, corresponding to a sound pressure level, can be received via a user interface. A stored lookup table can be accessed to convert the sound pressure level to a first product-specific logical volume level for each speaker. The stored lookup table can tabulate the non-standardized relationship between logical volume level and sound pressure level for each speaker. Data corresponding to the first product-specific logical volume level can be transmitted to each speaker.

Abstract: An equalization filter coefficient determinator continuously or quasi-continuously fades between a plurality of different equalizer settings in dependence on one or more setting parameters, to obtain a current set of equalization filter target coefficients describing a current equalizer setting. A number of setting parameters is smaller than a number of equalization filter target coefficients of current set of equalization filter target coefficients. An equalization filter coefficient determinator linearly combines a plurality of equalization filter target coefficient set components in dependence on one or more setting parameters. An equalization filter coefficient determinator obtains the current set of equalization filter target coefficients in dependence on position information obtained using a two-dimensional or three-dimensional user input device. An apparatus includes a user interface, an equalization filter coefficient determinator and an equalizer.

Abstract: Some embodiments of the invention are directed to enabling a user to easily identify the frequency range(s) at which sound masking occurs, and addressing the masking, if desired. In this respect, the extent to which a first stem is masked by one or more second stems in a frequency range may depend not only on the absolute value of the energy of the second stem(s) in the frequency range, but also on the relative energy of the first stem with respect to the second stem(s) in the frequency range. Accordingly, some embodiments are directed to modeling sound masking as a function of the energy of the stem being masked and of the relative energy of the masked stem with respect to the masking stem(s) in the frequency range, such as by modeling sound masking as loudness loss, a value indicative of the reduction in loudness of a stem of interest caused by the presence of one or more other stems in a frequency range.

Abstract: An apparatus includes a sensor for sensing an environment and a speaker. The apparatus also includes a display capable of displaying video information in multiple directions, wherein the display is configured to display video information in different directions based on information of an object sensed by the sensor. The speaker is configured to generate in multiple directions sound associated with the video information.

Abstract: Provided are, among other things, systems, methods and techniques for loudness-based audio-signal compensation. According to one such system, an input line accepts an input audio signal; a loudness estimator, coupled to the input line, processes the input audio signal to obtain an audio playback loudness level and also provides a production loudness level for the input audio signal; and a filter generator/selector coupled to an output of the loudness estimator provides an audio compensation filter based on the production loudness level and the audio playback loudness level. The input signal is processed by the audio compensation filter to provide an output signal.

Abstract: The present disclosure relates to a car audio information transmitter, including: a main controller, an audio receiver, an audio, transforming module, a wireless transmitter, and a frequency adjustment module. The audio receiver connects to the main controller. The audio transforming module connects to the audio receiver and is configured to transform the audio signals. The wireless transmitter connects to the audio transforming module and is configured to transmit the transformed audio signals to a wireless receiver of a car. The frequency adjustment module connects to the audio transforming module and is configured to adjust a frequency of the audio signals. The car audio information transmitter is capable of accessing to the adjusting mode by one operation and adjusting an amplitude of the audio signals. As such, the complicated operations may be avoided, and the driving safety may be improved.

Abstract: A method for setting a filter frequency response includes: dividing a target frequency band into multiple first partial bands at a frequency corresponding to an intersection between a first line representative of a target spectrum envelope and a second line representative of a first spectrum envelope derived from smoothing a measured frequency spectrum; and setting a frequency response of a filter corresponding to a first partial band specified by an operator from among the multiple first partial bands, in accordance with a setting operation for setting an adjustment amount performed by the operator with respect to the specified first partial band.

Abstract: A modular speaker system comprises a first speaker module; at least one second speaker module in communication with the first speaker module; and a controller that receives an instruction to selectively activate, reactivate, or inactivate the at least one second speaker module with respect to the first speaker module.

Abstract: A method for dynamic equalization is performed at a system. The system receives an electronic audio signal. The system monitors available headroom based on two or more of: an amplifier of the system, a change in volume of the system, characteristics of a transducer of a loudspeaker associated with the system, and the audio signal. Responsive to a change in the available headroom, the system modifies one or more parameters of an equalizer to at least a portion of the signal in order to extend a physical low-frequency response of the loudspeaker, the one or more parameters comprising a gain for the and a frequency.

Abstract: A microphone package for providing a modified microphone signal includes a microphone and an equalizer device coupled to the microphone.

Abstract: A coding apparatus includes a processor and a memory that stores instructions, which when executed causes the processor to perform operations, including encoding a first band of an input audio signal to be a first spectrum and dividing the first spectrum into a plurality of sub-bands. The operations also include searching a largest amplitude value of the divided first spectrum in each of the plurality of sub-bands, and normalizing the divided first spectrum in each of the plurality of sub-bands. The operations further include emphasizing a harmonic structure in the normalized first spectrum, and searching a best band that has a largest correlation value between each divided band of a second band spectrum and the emphasized first spectrum in which the harmonic structure is emphasized, and encoding the second band spectrum using lag information identifying the best band and transmitting the lag information to a decoder side.

Abstract: A method of providing audio information, said method comprising: (i) receiving audio filter settings in a client device; (ii) receiving audio data in the client device, wherein the received audio data is in an audio coding format; (iii) converting the audio filter settings to an audio filter signal in a processor of the client device, where the audio filter signal is a time-varying signal; (iv) converting the received audio data to an audio signal in a processor of the client device, where the audio signal is a time-varying signal; and (v) transmitting the converted audio filter signal and the converted audio signal from the client device to an audio output device, where the audio output device is separate from and in communication with the client device, and where the audio output device is configured for modifying the audio signal according to the audio filter signal to generate a time-varying audio output.

Abstract: An exemplary microphone capture simulation system accesses a captured set of audio signals captured by a plurality of directional microphones disposed at a plurality of locations on a perimeter of a capture zone of a real-world scene. The system identifies a location within the capture zone that corresponds to a virtual location at which a user is virtually located within a virtual reality space that is based on the capture zone. Based on the captured set of audio signals and the identified location, the system generates a simulated set of audio signals representative of a simulation of a full-sphere multi-capsule microphone capture at the identified location. The system processes the simulated set of audio signals to form a renderable set of audio signals configured to be rendered to simulate full-sphere sound for the virtual location while the user is virtually located at the virtual location.