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: Example methods and systems for modifying the playback of content using pre-processed profile information are described. Example instructions, when executed, cause at least one processor to access a media stream that includes media and a profile of equalization parameters, the media stream provided to a device via a network, the profile of equalization parameters included in the media stream selected based on a comparison of a reference fingerprint to a query fingerprint generated based on the media, the profile of equalization parameters including an equalization parameter for the media; and modify playback of the media based on the equalization parameter specified in the accessed profile.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization based on variant selection. An example apparatus to equalize audio includes at least one memory, machine readable instructions, and processor circuity to at least one of instantiate or execute the machine readable instructions to train a neural network model to apply a first audio equalization profile to first audio associated with a first variant of media, and apply a second audio equalization profile to second audio associated with a second variant of media. The processor circuitry is to at least one of instantiate or execute the machine readable instructions to at least one of dispatch or execute the neural network model.

Abstract: A voice output from an i-th audio source apparatus is adjusted with a frequency transfer function Wii(f) set in an i-th noise reduction filter and is output from an i-th speaker. In the noise reduction filter, a frequency transfer function Wii(f) is set in which the sound is made smaller at frequencies where the gain of the frequency transfer function Cim(f) from the SPKi to a user tends to be relatively larger than the gain of the frequency transfer function Cii(f) from the SPKi to the user, and the sound is increased at frequencies where the gain of Cim(f) tends to be relatively smaller than the gain of Cii(f), where m is an integer excluding i from 1 to n.

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: A device, method, and medium for integrating monaural and binaural beats into music are provided. The music is analyzed to determine key, root tone, and spectral range. It is then remixed with monaural beats and/or binaural beats at frequencies based on a desired entrainment frequency and the root tone and lowest dominant frequency range of the music. Additional harmonics of the beats in higher octaves may be integrated into the music as well using mixing and/or equalization.

Abstract: A sound crosstalk suppression device includes: a speaker analysis unit configured to analyze a speaker situation in a closed space based on voice signals respectively collected by a plurality of microphones arranged in the closed space; a filter update unit that includes a filter configured to generate a suppression signal of a crosstalk component included in a voice signal of a main speaker, that is configured to update a parameter of the filter, and that is configured to store the updated parameter in a memory; a reset unit configured to reset the parameter of the filter in a case where it is determined that an analysis result of the speaker situation is switched; and a crosstalk suppression unit configured to suppress a crosstalk component by using a suppression signal.

Abstract: Systems and method are disclosed for facilitating efficient calibration of filters for correcting room and/or speaker-based distortion and/or binaural imbalances in audio reproduction, and/or for producing three-dimensional sound in stereo system environments. According to some embodiments, using a portable device such as a smartphone or tablet, a user can calibrate speakers by initiating playback of a test signal, detecting playback of the test signal with the portable device's microphone, and repeating this process for a number of speakers and/or device positions (e.g., next to each of the user's ears). A comparison can be made between the test signal and the detected signal, and this can be used to more precisely calibrate rendering of future signals by the speakers.

Abstract: Methods and systems for performing automatic speed-based audio control. One method includes receiving, with an electronic control unit included in a vehicle, a speed of the vehicle and receiving, with the electronic control unit, an audio signal. The method also includes accessing, with the electronic control unit, a plurality of equalization curves based on the speed of the vehicle, each of the plurality of equalization curves associated with the speed of the vehicle and each of the plurality of equalization curves defining a gain adjustment for one of a plurality of frequencies, and, for each curve of the plurality of equalization curves, applying the gain adjustment defined by the curve to one of the plurality of frequencies of the audio signal.

Abstract: An example implementation involves a computing device transmitting, via a local area network, a command that instructs a playback device to play a particular audio signal. The example implementation also involves the computing device receiving data indicating a detected audio signal corresponding to playback of the particular audio signal by the playback device, where the detected audio signal includes a portion of the particular audio signal. The implementation further involves the computing device obtaining data indicating a predetermined audio characteristic and determining an audio processing algorithm based on the detected audio signal and the predetermined audio characteristic. The example implementation involves causing the playback device to apply the determined audio processing algorithm when playing audio via at least one speaker.

Abstract: Disclosed are methods, systems, and non-transitory computer-readable storage media for reproducing music to mimic a live performance. The present technology includes means for receiving input corresponding to a plurality of component audio files associated with an original audio file; determining audio profiles of the plurality of component audio files and frequency profiles of a plurality of audio generators; determining a routing scheme for the plurality of component audio files and the plurality of audio generators, wherein the routing scheme is based on the audio profiles and the frequency profiles; and generating an output based on the routing scheme.

Abstract: An electronic device and method for recommendation of audio based on video analysis is provided. The electronic device receives one or more frames of a first scene of a plurality of scenes of a video. The first scene includes a set of objects. The electronic device applies a trained neural network model on the received one or more frames to detect the set of objects. The electronic device determines an impact score of each object of the detected set of objects of the first scene based on the application of the trained neural network model on the set of objects. The electronic device further selects at least one first object from the set of objects based on the impact score of each object, and recommends one or more first audio tracks as a sound effect for the first scene based on the selected at least one first object.

Abstract: Audio device apparatuses for hearing impaired users comprise a wearable housing connected, wired or wirelessly, to earphones or a headset. The audio device receives incoming sound, processes the incoming sound to form processed sound, and delivers the processed sound to the earphones or headset of the user.

Abstract: An example method of operation may include identifying, in a particular room environment, a number of speakers and one or more microphones on a network controlled by a controller and amplifier, providing test signals to play sequentially from each amplifier channel of the amplifier and the speakers, monitoring the test signals from the one or more microphones simultaneously to detect operational speakers and amplifier channels, providing additional test signals to the speakers to determine tuning parameters, detecting the additional test signals at the one or more microphones controlled by the controller, and automatically establishing a background noise level and noise spectrum of the room environment based on the detected additional test signals.

Abstract: Systems, methods, and devices relate to tunable circuits for multimode power amplification. For example, a variable-gain amplification system can include an amplifier configured to provide an amplified signal and to selectively operate in at least a first gain mode and a second gain mode. The variable-gain amplification system can also include a tunable circuit configured to receive the amplified signal from the amplifier and to provide an output signal. The tunable circuit can be configured to implement a first turn ratio for the first gain mode and a second turn ratio for the second gain mode.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization. Example instructions disclosed herein cause one or more processors to at least: detect an irregularity in a frequency representation of an audio signal in response to a change in volume between a set of frequency values exceeding a threshold; and adjust a volume at a first frequency value of the set of frequency values to reduce the irregularity.

Abstract: Systems and methods are disclosed for audio enhancement. For example, methods may include accessing audio data; determining a window of audio samples based on the audio data; inputting the window of audio samples to a classifier to obtain a classification, in which the classifier includes a neural network and the classification takes a value from a set of multiple classes of audio; selecting, based on the classification, an audio enhancement network from a set of multiple audio enhancement networks; applying the selected audio enhancement network to the window of audio samples to obtain an enhanced audio segment, in which the selected audio enhancement network includes a neural network that has been trained using audio signals of a type associated with the classification; and storing, playing, or transmitting an enhanced audio signal based on the enhanced audio segment.

Abstract: A method comprises outputting at least one test signal to a listening environment via a loudspeaker, receiving each of the at least one test signal via a microphone arranged in the listening environment, determining at least one of a position and an orientation of the microphone in the listening environment with respect to the loudspeaker at the time of receipt of each of the at least one test signal, evaluating the test signals received by the microphone, and determining at least one aspect related to at least one transfer function based on the evaluation of the test signal and on at least one of the position and the orientation of the microphone.

Abstract: An apparatus configured to: obtain at least one spatial audio signal that defines an audio scene forming at least in part an immersive media content; obtain metadata associated with the at least one spatial audio signal; obtain at least one augmentation control parameter associated with the at least one spatial audio signal; obtain at least one augmentation audio signal; render an output audio signal that is based, at least partially, on the at least one spatial audio signal, the metadata associated with the at least one spatial audio signal, the at least one augmentation control parameter, and the at least one augmentation audio signal; and obtain an indication that at least part of the at least one spatial audio signal has been omitted from the output audio signal based, at least partially, on at least part of the at least one augmentation audio signal included in the output audio signal.

Abstract: To encourage viewer participation, games, contests and social interactions are able to be synchronized with programming such as television shows or commercials utilizing a second screen such as a cell phone, iPad® or laptop computer. The programming is able to be television programming, Internet programming (e.g. a video displayed on a webpage or mobile device) or any other programming. The gaming is able to be any game such as a game of skill or chance, for example, a scavenger hunt or a treasure hunt.

Abstract: Provided is a direction of arrival estimation device wherein: a calculation circuit calculates a frequency weighting factor for each of a plurality of frequency components of signals recorded in a microphone array, on the basis of the differences among unit vectors indicating the directions of the sound sources of each of the plurality of frequency components; and an estimation circuit estimates the direction of arrival of a signal from the sound source, on the basis of the frequency weighting factors.

Abstract: An audio processing device includes an input terminal to input an audio signal and a peak shift filter. The peak shift filter increases sound pressure with a first center frequency in the input audio signal as a first peak, and shifts the first center frequency between a preset lowest frequency and a preset highest frequency.

Abstract: Processes and devices for equalizing an audio system that is adapted to use a loudspeaker to transduce test audio signals into test sounds. The processes and devices can involve the use of infrared signals to convey information in one or both directions between the audio system and a portable computer device that captures test sounds, calculates audio parameters that can be used in the equalization process, and transmits these audio parameters back to the audio system for its use in equalizing audio signals that are played by the audio system.

Abstract: The disclosure relates to an electronic apparatus for recognizing user voice and a method of recognizing, by the electronic apparatus, the user voice. According to an embodiment, the method of recognizing the user voice includes obtaining an audio signal segmented into a plurality of frame units, determining an energy component for each filter bank by applying a filter bank distributed according to a preset scale to a frequency spectrum of the audio signal segmented into the frame units, smoothing the determined energy component for each filter bank, extracting a feature vector of the audio signal based on the smoothed energy component for each filter bank, and recognizing the user voice in the audio signal by inputting the extracted feature vector to a voice recognition model.

Abstract: A micro electro-mechanical film speaker device is provided. A signal input terminal of a signal amplifier circuit is connected to an external input circuit. A second output terminal of the signal amplifier circuit is connected to a negative terminal of a micro electro-mechanical film speaker. A first terminal of a first high-pass filter is connected to a first output terminal of the signal amplifier circuit. A second terminal of the first high-pass filter is connected to a positive terminal of the micro electro-mechanical film speaker. A first terminal of a second high-pass filter is connected to a feedback terminal of the signal amplifier circuit. A second terminal of the second high-pass filter is connected to the positive terminal of the micro electro-mechanical film speaker.

Abstract: The application describes techniques for adaptively setting a threshold that will be used to determine the gain applied to a haptics signal.

Abstract: A control device includes: a difference calculator that calculates a maximum gain difference between frequency characteristics of an audio signal and desired frequency characteristics in a predetermined frequency band; a center frequency calculator that calculates a center frequency between a first frequency at which the gain difference is 1/?2 times the maximum gain difference and which is closest to a peak frequency corresponding to the maximum gain difference among one or more frequencies lower than the peak frequency and a second frequency at which the gain difference is 1/?2 times the maximum gain difference and which is closest to the peak frequency among one or more frequencies higher than the peak frequency; a determiner that determines parameters of an equalizer based on the maximum gain difference and the center frequency; and an outputter that outputs the parameters to the equalizer.

Abstract: The application relates to a sound quality output method. The method includes: obtaining the current volume level input by the user; determine the audio signal gain difference between the current volume level and the baseline volume level according to the current volume level and the preset baseline volume level, determining an equal loudness curve difference between the current volume level and the baseline volume level according to the audio signal gain difference, and the current volume level and the baseline volume level, determining the audio quality response curve corresponding to the current volume level according to the equal loudness curve difference and the pre-stored audio quality response curve corresponding to the baseline volume level, determining the output signal parameters of multiple audio output channels according to the audio quality response curve corresponding to the current volume level. The application optimizes the sound quality effect and enhances acoustic experience.

Abstract: A full-range speaker is driven by an original sound signal output from a sound source. A vibration detection unit detects vibration of the full-range speaker and outputs a playback signal. A subtractor outputs an error signal indicating the difference between the original sound signal and the play back signal. The error signal is amplified by the amplifier to drive the woofer which increases the sound pressure level in the low frequency range by outputting a sound in the same phase as the sound of the full range speaker if the sound pressure level of the full range speaker is insufficient, or in the opposite phase as the sound of the full range speaker if the sound pressure level is excessive. If the sound pressure level is excessive, the sound pressure level is reduced by outputting a sound in phase with the sound of the full range speaker.

Abstract: An audio device includes a first earbud and a second earbud, each of which is configured to establish respective first and second wireless links with an audio source, and receive, from the audio source, audio information over the respective first and second wireless links. The first earbud and the second earbud are further configured to communicate with one another over a third wireless link. At least one of the first earbud and the second earbud includes a measurement circuit configured to measure respective signal strengths of the first, second and third wireless links, a computation circuit configured to compute a difference in signal strengths of the first and second wireless links, and a determination circuit configured to determine an operating environment of the audio device based on the computed difference and further based on the signal strength of the third wireless link.

Abstract: The present invention is a system and a method for processing and adjusting the PCM digital audio signals using specific reverberation and equalization settings that have been determined to potentially improve certain physical health parameters measurements, as determined by conducting bio-signal testing. The system includes a source of audio signals, producing an analog audio signal as input; a digital to analog converter, converting the analog audio signal to digital; a digital system processor, having a computer processor and memory or circuitry for processing the input audio signal using an equalization, a reverberation and a volume setting that is measured to produce an audio output signal that has more beneficial health response on human physiological functions than an unprocessed PCM digital signal or a base measurement without any audio signal, as measured by at least one bio-sensor attached to at least one listener.

Abstract: An embodiment of the present invention may be deployed in a system comprising a media player and a remote server operably connected to communicate with one another. The invention allows for a settings file to be stored on the remote server, the settings file comprising parameters useful for adjusting different settings on the media player such that rendering of a particular content to be made possible on the media player according to a user's predetermined taste by downloading the settings from the server onto the media player.

Abstract: Example techniques may involve headphone interaction. An implementation may include while headphones are disconnected from a control device, the control device receiving an indication of particular audio content being played back by a first zone of the media playback system. While the particular audio content is being played back by the one or more playback devices of the media playback system, detecting that headphones have been connected to the first control device. Headphones may be connectable to the control device via either (a) an analog headphone jack or (b) a point-to-point personal area network connection. In response to detecting that the headphones are connected to the control device: the control device (i) causing the one or more playback devices of the first zone to stop playback of the particular audio content; (ii) retrieving the particular audio content and (iii) resuming playback of the particular audio content via the connected headphones.

Abstract: The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

Abstract: An audio playback device comprises a microphone, a speaker, and a processor. The processor is arranged to output by the speaker first audio content and receive by the microphone an indication of the first audio content. A first acoustic response of a room in which the audio playback device is located is determined based on the received indication of first audio content. A mapping is applied to the first acoustic response to determine a second acoustic response. The second acoustic response is indicative of an approximated acoustic response of the room at a spatial location different from a spatial location of the microphone. The second audio content output by the speaker is adjusted based on the second response.

Abstract: An audio-enhanced furniture system comprises: a furniture assembly; an upholstery fabric at least partially covering the furniture assembly; and a speaker system positioned within the furniture assembly, the speaker system including a speaker covered by the upholstery fabric. The speaker is configured to be tuned to compensate for sound being emitted from the speaker through the upholstery fabric.

Abstract: Loudspeakers are described that may reduce comb filtering effects perceived by a listener by either 1) moving transducers closer to a sound reflective surface (e.g., a baseplate, a tabletop or a floor) through vertical (height) or rotational adjustments of the transducers or 2) guiding sound produced by the transducers to be released into the listening area proximate to the reflective surface through the use of horns and openings that are at a prescribed distance from the reflective surface. The reduction of this distance between the reflective surface and the point at which sound emitted by the transducers is released into the listening area may lead to shorter reflected path that reduces comb filtering effects caused by reflected sounds that are delayed relative to the direct sound. Accordingly, the loudspeakers shown and described may be placed on reflective surfaces without severe audio coloration caused by reflected sounds.

Abstract: A vehicle and a controlling method of the same include a speaker outputting virtual engine sound integrated with a head lamp. The vehicle includes the head lamp; a head lamp case in which the head lamp is provided; and a speaker for outputting a virtual engine sound, and provided inside the head lamp case, where an internal cross-sectional area of the head lamp case may increase in a direction toward a front of the vehicle based on a position where the speaker is provided.

Abstract: An example method of operation may include determining a frequency response to a measured chirp signal detected from one or more speakers, determining an average value of the frequency response based on a high limit value and a low limit value, subtracting a measured response from a target response, and the target response is based on one or more filter frequencies; determining a frequency limited target filter with audible parameters based on the subtraction, and applying an infinite impulse response (IIR) biquad filter based on an area defined by the frequency limited target filter to equalize the frequency response of the one or more speakers.

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: There is provided a method for controlling bass reproduction properties of a multichannel audio system, wherein the audio system has inputs for at least two audio input signals and includes a set of loudspeakers, including at least one bass-capable loudspeaker and at least two high-range loudspeakers, each loudspeaker being associated with a loudspeaker channel. The method includes obtaining impulse responses or transfer functions that represent the sound reproduction properties of each loudspeaker channel at a number of measurement or control positions. The method also includes tuning, when the audio system includes more than one bass-capable loudspeaker, loudspeaker channels of at least two bass loudspeakers to each other so that their sum impulse response has minimum spatial variability, and/or controlling high-range loudspeaker speaker channels to be in-phase with each other and/or with bass-capable loudspeaker channel in a crossover frequency band.

Abstract: An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

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: Systems and methods utilize a modified genetic algorithm for adapting an off-the-shelf audio system, such as in a high-end television, to a given, particular room or other physical location presenting a specific or unique auditory environment with a set of acoustic properties. An audio system is adapted to a given room by determining an IIR based EQ solution via iterative techniques, including an iterative technique based upon a genetic algorithm adapted for an audio frequency response equalization application. In a variant, an audio system is adapted to a particular room, adjust the EQ across a microphone's bandwidth while preserving the factory-calibrated EQ response across the remaining bandwidth.

Abstract: An electronic device includes a balloon configured to contact a surface of a human conduit and one or more a biometric sensors operatively coupled to or on or in or within the balloon for detecting a biometric signal. The electronic device in some examples is a biometric sensor or measuring device. In other examples, the electronic device is an integrated device such as an earpiece having biometric sensors. In yet other examples, the electronic device can be operatively coupled to other device and other biometric sensors. Other embodiments are disclosed.

Abstract: Example techniques involve a calibration state variable. An example implementation receives, via a network interface, an indication that the first playback device is calibrated. Based on receiving the indication that the first playback device is calibrated, the example implementation updates a calibration state variable to indicate that the first playback device is calibrated, wherein the calibration state variable is stored in the data storage. The example implementation sends, via the network interface, an indication of the updated calibration state variable to a second device.

Abstract: A radar sensor system is provided. The radar sensor system includes: at least two radar sensors each having at least one transmitter and at least one receiver, detection regions of the two radar sensors overlapping at least partially. The two radar sensors are situated at a defined distance from one another. Transmit signals of the two radar sensors are synchronizable in such a way that radiation of one radar sensor that was emitted by the respective other radar sensor and reflected by an object is capable of being evaluated by an evaluation device.

Abstract: The present invention discloses a method for producing parametric sound using parametric sound system which is based on ultrasonic electrostatic transducers. It comprises modulation of a carrier ultrasonic signal with a processed audio signal in audio signal processor comprising adaptive frequency filtering based on the audio signal level, dynamic range compression, square root operation, amplification of the modulated ultrasonic signal using a D-class amplifier, driving an electrostatic transducer and generating modulated ultrasonic waves into the air. The electrostatic transducer for the parametric sound system comprises a specific back plate structure that improves electromechanical efficiency of the transducer and also enables realization of a phased array on a single back plate. The disclosed manufacturing method of the electrostatic transducer comprises producing sets of electrodes on the surface of the back plate forming individual cells.

Abstract: An example method of operation may include identifying, in a particular room environment, a number of speakers and one or more microphones on a network controlled by a controller and amplifier, providing test signals to play sequentially from each amplifier channel of the amplifier and the speakers, monitoring the test signals from the one or more microphones simultaneously to detect operational speakers and amplifier channels, providing additional test signals to the speakers to determine tuning parameters, detecting the additional test signals at the one or more microphones controlled by the controller, and automatically establishing a background noise level and noise spectrum of the room environment based on the detected additional test signals.

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.