Abstract: An audio system includes at least one sound generation device having at least an acoustic transducer; and an electronic integrated circuit. A communication bus is connected to the at least one sound generation device and is configured to communicate a digital signal comprised of one or more audio streams and control signals. A power bus is connected to the at least one sound generation device. The acoustic transducer includes at least one of the following groups: a membrane and an acoustic modulator; or a membrane, acoustic resonator, and acoustic coupler. The electronic integrated circuit is constructed and arranged to receive the digital signal and generate an analog electric signal to operate the acoustic transducer to generate an audio signal in accordance with the control signal.

Abstract: An audio signal input and output device includes a port that inputs or outputs an audio signal, an interface that receives a specification of a channel or bus to be assigned to the port, and a sender that sends information over a network for assigning the channel or bus based on the specification received by the interface. The information assigns the inputted audio signal to a predetermined input channel of the management device. A speaker emits a sound based on the inputted audio signal.

Abstract: A hearing device configured for audio communication is disclosed. The hearing device comprises a wireless communication unit and a processing unit, the wireless communication unit being configured to receive and transmit audio signals. The processing unit is configured to receive and/or transmit the audio signal from/to the wireless communication unit, process the audio signal in a first frequency band or in a second frequency band, and detect the quality of the audio communication. If the audio signal is processed in the first frequency band, the processing unit is switching to processing in the second frequency band if the quality of the audio communication is below a first predetermined threshold. If the audio signal is processed in the second frequency band, the processing unit is switching to processing in the first frequency band if the quality of the audio communication is above the first predetermined threshold.

Abstract: A method for controlling an FIR filter includes generating, based on operation information, a first and second control data, generating a second amplitude characteristic, and setting filter coefficients of the FIR filter based on the second amplitude characteristic. The first control data indicates an amount of expansion/compression in a frequency axis direction of a first amplitude characteristic that corresponds to a predetermined transfer function that is expressed as a function of an angular frequency, and the amount of expansion/compression is an integer value or a non-integer value. The second control data indicates an amount of shift in the frequency axis direction of the first amplitude characteristic.

Abstract: The invention relates to a method for processing an acoustic input signal, preferably a speech signal, said method comprising the following steps: a) receiving a digital representation (Sin) of an acoustic input signal, b) calculating at least one statistical parameter (P) of the digital representation (Sin) of the acoustic input signal, c) calculating a compression ratio function (CRf) based—on a prescribed constant compression ratio (CRpr), said prescribed constant compression ratio (CRpr) uniformly mapping acoustic input signals of a selected magnitude to acoustic output signals of a selected magnitude, and—on at least one statistical parameter (P) calculated in step b), and d) applying the non-uniform compression ratio function (CRf) according to step c) on the digital representation (Sin) of the acoustic input signal delivering a digital representation (Sout) of an enhanced acoustic output signal.

Abstract: An information processing device includes a first time-frequency converter configured to perform a time-frequency conversion with respect to an input signal, using a window function having a first width, a second time-frequency converter configured to perform a time-frequency conversion with respect to the input signal, using a second window function having a second width smaller than the first width, and a modification processing unit configured to modify an output of the second time-frequency converter, using a frequency analysis result based on an output of the first time-frequency converter.

Abstract: Disclosed are a device and method for wirelessly communicating. The device according to one example embodiment of the present disclosure may comprise a transceiver and a controller connected to the transceiver, wherein the controller is configured to identify at least one additional sample on the basis of a digital signal by using a neural network model and upscale the digital signal by adding the at least one identified additional sample to a plurality of samples of the digital signal.

Abstract: Disclosed are a sound source output apparatus applying a plurality of DACs and an operating method thereof. The sound source output apparatus according to the exemplary embodiment of the present disclosure includes a sound source acquiring unit which acquires a sound source signal; a tag identification processing unit which identifies tag information about the sound source signal; a DAC selection control unit which selects a specific DAC among a plurality of digital analog converters (DACs) based on a tag identification result of the tag information or a user input signal; and a sound source output unit which outputs a sound source which is converted by the selected specific DAC.

Abstract: Provided a robot for assisting hearing of a user, while minimizing an influence on the surroundings. The robot includes a speaker, a microphone configured to recognize a voice, a processor configured to acquire a position of a user's face when a hearing aid command is acquired on the basis of the voice recognized through the microphone, and a driving unit configured to cause the speaker to be moved toward the position of the user's face, wherein the processor acquires a sound which is a target of hearing aid, generates an assistant sound by amplifying a predetermined frequency band of the sound or converting the predetermined frequency band of the sound into a different frequency band, and outputs the assistant sound through the speaker.

Abstract: A method performed by an in-ear headphone that includes a speaker and an internal microphone. The method receives a microphone signal from the internal microphone that indicates a current sound pressure level (SPL) in an ear canal of a user. The current SPL is a result of a control leak from the in-ear headphone into an ambient environment that reduces a SPL in the ear canal between 2 dB and 25 dB at a frequency within a frequency range than if otherwise not present. The method determines an active noise cancellation (ANC) filter based on the microphone signal and generates an anti-noise signal using the ANC filter. The method drives the speaker using the anti-noise signal to reduce the current SPL in the ear canal of the user as much as 25 dB at a frequency within the frequency range.

Abstract: A method for testing a multiple-actuator storage device includes accessing a first storage device comprising at least a first and a second logical unit and concurrently performing a first group of test operations on the first logical unit and a second group of test operations on the second logical unit. The method includes monitoring for an interruption during the performing the group of test operations on the first or the second logical unit, and during the monitoring, if it is determined that there is an interruption on one of the first or second logical units, performing the respective test operations on the other logical unit is temporarily suspended. The method also includes continuing to monitor for the interruption, and upon determining that the interruption is not currently active, resuming concurrently performing the test operations on the first and second logical units.

Abstract: A headphones system includes reference headphones having native audio performance characteristics, and a personal communication device that generates a user interface presenting visual representations of other headphones. In response to user selection of a visual representation of particular headphones, the personal communication device identifies a corresponding current sound profile establishing a mapping between audio performance characteristics of the particular user selected headphones and the native audio performance characteristics, and communicates the current sound profile to the reference headphones.

Abstract: The embodiments of the present disclosure disclose a method and device for processing an audio signal. A specific embodiment of the method includes acquiring a head image of a target user and an audio signal to be processed; determining head attitude angles of the target user based on the head image, and determining a distance between a target sound source and the head of the target user; and inputting the head attitude angles, the distance and the audio signal to be processed into a preset head related transfer function to obtain a processed left channel audio signal and a processed right channel audio signal, wherein the head related transfer function is used to characterize a correspondence between the head attitude angles, the distance and the audio signal to be processed, and the processed left channel audio signal and the processed right channel audio signal.

Abstract: There is disclosed a device that can record a sound by selecting a directivity direction as desired by the user at the time of recording and can play back the sound by selecting, at the time of playback, a desired directivity direction by the user, separate from the directivity direction specified at the time of recording. The device has microphones, a touch panel for specifying a directivity direction, and a signal processing unit. A recording module performs directivity control in a direction specified on the touch panel and records audio data subjected to effect processing in a memory as directivity-controlled data, and records audio data that are not subjected to directivity control or effect processing in the memory as directivity-control unprocessed data.

Abstract: Disclosed are a method for detecting speech concatenating points and a storage medium. The method includes: acquiring a speech to be detected, and determining high-frequency components and low-frequency components of the speech to be detected; extracting first cepstrum features and second cepstrum features corresponding to the speech to be detected according to the high-frequency components and the low-frequency components; splicing the first and the second cepstrum feature of speech per frame in the speech to be detected in units of frame so as to obtain a parameter sequence; inputting the parameter sequence into a neural network model so as to obtain a feature sequence corresponding to the speech to be detected, wherein the model has been trained, has learned and stored a correspondence between the parameter sequence and the feature sequence; and performing detection of speech concatenating points on the speech to be detected according to the feature sequence.

Abstract: This disclosure provides methods, systems, and apparatuses, for a microphone circuit. In particular, the circuit includes transducer that can sense pressure changes and generate an electrical signal having frequency components in a first frequency range and in a second frequency range higher than the first frequency range. The circuit includes a feedback circuitry that can attenuate frequency components in the first frequency range in the electrical signal and, from it, generate an audio signal. A feedback path circuit includes a low pass filter having a cut-off frequency within the first frequency range, and filters the audio signal to generate a low pass filter signal that includes frequency components in the first frequency range. The low pass filter signal can be used to generate a low frequency pressure signal that corresponds to low frequency pressure changes sensed by the transducer.

Abstract: A computer implemented method for managing a sound emitting device comprising: receiving data associated with operation of the sound emitting device at a predetermined location; processing said data to determine an operating characteristic of that device for that location; comparing the operating characteristic with a predetermined mathematical relationship to determine whether a difference exists; and identifying an input adjustment to correct the difference wherein the input adjustment optionally is within a predetermined range and optionally does not exceed a predetermined maximum increment; wherein the predetermined mathematical relationship is between an input variable and an output variable in respect of the sound emitting device.

Abstract: A method for creating an output soundfield signal from an input soundfield signal, the method including the steps of: (a) forming at least one delayed signals from the input soundfield signal, (b) for each of the delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and (c) combining together the acoustically transformed delayed signals and the input soundfield signal to produce the output soundfield signal.

Abstract: The present technology relates to an information processing apparatus and method, and a program that enable more efficient edit. The information processing apparatus includes a control unit that selects and groups a plurality of objects existing in a predetermined space, and changes the positions of the plurality of the objects while maintaining the relative positional relationship of the plurality of the grouped objects in the space. The present technology can be applied to information processing apparatuses.

Abstract: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: A signal processor for a sound reproduction system which is arranged to perform processing of a sound recording so as to provide input signals for an array of distributed loudspeakers, such that the sound field generated by the loudspeakers results in cross-talk cancellation in respect of multiple listener positions at substantially all frequencies reproduced by the loudspeakers, and wherein the sound field so generated is an approximation of a sound field produced by an Optimal Source Distribution, OSD.

Abstract: A system and method are disclosed for ignoring a wakeword received at a speech-enabled listening device when it is determined the wakeword is reproduced audio from an audio-playing device. Determination can be by detecting audio distortions, by an ignore flag sent locally between an audio-playing device and speech-enabled device, by and ignore flag sent from a server, by comparison of received audio played audio to a wakeword within an audio-playing device or a speech-enabled device, and other means.

Abstract: An audio processing apparatus includes a linear processing stage and control signal circuitry (CSC). The linear processing stage is configured to receive an input audio signal, and to linearly process the input audio signal based on a user-provided setting of the linear processing stage, so as to produce a linear stage output audio signal. The control signal circuitry (CSC), which is configured to (a) generate a control signal based on (i) the user-provided setting of the linear stage and (ii) a prespecified signal quality preserving criterion, and (b) control the linear stage output audio signal with the control signal, so as to produce a controlled audio signal, in compliance with the signal quality preserving criterion.

Abstract: The present disclosure provides a method and an apparatus for simulating the sound characteristics of the audio processor, a terminal device and a storage medium. The method simulates the sound characteristic of a target audio processor by constructing a nonlinear model. The nonlinear model consists of a pre-filter, a saturation characteristic curve, and a post-filter. A high-level signal is sent to the target sound processor, and the target sound processor correspondingly sends out the first output signal, and then the first output signal is filtered and performed spectrum analysis to acquire the frequency response of the post-filter; And a low-level signal is sent to the target sound processor, and the target sound processor correspondingly sends out the second output signal, and then the second output signal is filtered and performed spectrum analysis to obtain a frequency response product of the pre-filter and the post-filter.

Abstract: A method, system and storage medium for solving echo cancellation failure; method comprises: starting echo cancellation algorithm for convergence of self-adapting filter to stabilize coefficient of filter function (S100); presetting maximum amplitude value for signal intensity that microphone can receive, and detecting if microphone signal lower than maximum amplitude value (S200); prompting user or television system automatically to turn down television volume when microphone received signal larger than or equal to maximum amplitude value, to ensure microphone received signal intensity lower than maximum amplitude value (S300); echo cancellation being carried out without distortion of signals, wherein coefficient of self-adapting filter function W(n) may be continually kept stable, avoiding self-adapting filter converging multiple times due to signal distortion (S400).

Abstract: At least one exemplary embodiment is directed to a communication device that includes a microphone configured to detect an acoustic signal from an acoustic environment, and a processor, configured to detect an acoustical dampening between the acoustic environment and the microphone, based on a change in a characteristic of the acoustic signal and, responsive to the acoustical dampening, apply a compensation filter to the acoustic signal to form a compensated acoustic signal that is reproduced. Other embodiments are disclosed.

Abstract: Noise reduction in a robot system includes the use of a gesture library that pairs noise profiles with gestures performed by the robot. A gesture to be performed by the robot is obtained, and the robot performs the gesture. The robot's performance of the gesture creates noise, and when a user speaks to the robot while the robot performs a gesture, incoming audio includes both user audio and robot noise. A noise profile associated with the gesture is retrieved from the gesture library and is applied to remove the robot noise from the incoming audio.

Abstract: A language proficiency analyzer automatically evaluates a person's language proficiency by analyzing that person's oral communications with another person. The analyzer first enhances the quality of an audio recording of a conversation between the two people using a neural network that automatically detects loss features in the audio and adds those loss features back into the audio. The analyzer then performs a textual and audio analysis on the improved audio. Through textual analysis, the analyzer uses a multi-attention network to determine how focused one person is on the other and how pleased one person is with the other. Through audio analysis, the analyzer uses a neural network to determine how well one person pronounced words during the conversation.

Abstract: Method for carrying out a morphing process, wherein an output parameter relating to the output of an audio signal outputted into an interior via an audio output device is changed.

Abstract: A method for determining a configuration parameter includes: transmitting a detection signal; receiving a reflection signal formed after the detection signal is reflected by a characteristic structure of an ear of a user; adjusting a signal transmission parameter of the detection signal according to a signal difference between the reflection signal and a standard signal until the signal difference meets a first preset condition, to determine the configuration parameter of an earphone.

Abstract: A method for compensating for acoustic influence of a listening room on an acoustic output from an audio system including at least a left and a right loudspeaker, the method comprising determining a left frequency response and a right frequency response, designing a left compensation filter FL, and a right compensation filter FR, and during playback applying the left and right filters to left and right channel inputs. According to the invention, a target response in the listening position is simulated, and the left and right compensation filters are designed to filter transfer functions based on the simulated target function multiplied by an inverse of the left/right frequency responses. By relying on a simulated target instead of relying on an empirical approach, the general impact of a room can be more accurately captured by the target functions.

Abstract: A method of operating a UI for controlling a virtual musical instrument can include receiving a first input corresponding to a selection of an array of cells within a cell matrix, each array of the cell matrix assigned to audio sample data stored in a computer-readable medium that, when triggered, causes the audio sample data to output corresponding audio, and each cell within its particular array of the cell matrix, in response to being selected for playback and upon being triggered to begin playback, causes the audio sample data corresponding to the cell's particular array to be played. The method can further include receiving a second input corresponding to a changing of a number of cells within the particular selected array; and changing the number of cells within the selected array based on the second input and maintaining a number of cells in other arrays within the cell matrix.

Abstract: Provided is an electronic device. The electronic device may include: a housing including a front plate, a rear plate facing away from the front plate, and a side member surrounding a first space between the front plate and the rear plate; a display panel exposed to an exterior of the electronic device through the front plate and configured to detect a pen input using a magnetic field; a speaker structure disposed between the display panel and the rear plate, and including a first surface facing the display panel, a second surface facing in a direction opposite the display panel, a side surface surrounding a second space between the first surface and the second surface, and a yoke facing the first surface; and a first shield structure made of a ferromagnetic material and disposed on the first surface.

Abstract: The present application discloses a speaker earphone. Each earphone unit includes a first speaker group and/or a second speaker group. The first speaker group includes a first speaker, a second speaker, and at least one capacitor. The second speaker group includes a first speaker, a second speaker, a third speaker, and at least one capacitor. In the present disclosure, at least two speakers are arranged on an enclosure of the earphone. The two speakers are in reverse connection according to polarity in order to achieve a better sound effect.

Abstract: A method, and system, of digital room correction for a device, such as a smart speaker, including a loudspeaker. The method comprises capturing audio from an environment local to the device, for example from one or more microphones of a smart speaker. The captured audio is then processed to recognize one or more categories of sound. A digital room correction procedure may then be controlled dependent upon recognition and/or analysis of at least one of the categories of sound.

Abstract: The present disclosure relates to near-end crosstalk (NEXT) cancellation. A transmit communication signal is transmitted over a first Digital Subscriber Line (DSL) connection using a first group of frequencies and a receive communication signal is received over a second DSL connection using a second group of frequencies that at least partially overlaps the first group of frequencies. A crosstalk correlation between the first and second communication signals is determined. Based on the crosstalk correlation, a crosstalk cancellation signal is generated. The crosstalk cancellation signal is subtracted from the second communication signal, with the intention of reducing NEXT.

Abstract: A system includes a circuitry that provides for psychoacoustic frequency range extension for a speaker. The circuitry generates quadrature components from an audio channel, and generates rotated spectral quadrature components by applying a forward transformation that rotates a spectrum of the quadrature components from a standard basis to a rotated basis. In the rotated basis, the circuitry isolates components of the rotated spectral quadrature components at target frequencies, and generates weighted phase-coherent harmonic spectral quadrature components by applying a scale-independent nonlinearity to the isolated components. The circuitry generates a harmonic spectral component by applying an inverse transformation that rotates a spectrum of the weighted phase-coherent harmonic spectral quadrature components from the rotated basis to the standard basis.

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: Embodiments relate to an audio processing system that adjusts for unmatched speakers, such as by frequency response, output power, directionality, etc. For example, a mobile device may include an earpiece speaker and a micro-loudspeaker. The audio processing system may include a crossover network that separates an input audio signal into a low frequency signal and a high frequency signal. Subband spatial processing and b-chain processing are applied to the high frequency signal to spatially enhance the input signal, and adjust the input signal for the unmatched speakers. The low frequency signal is processed using a parametric band-pass filter and a first gain to generate a low frequency resonator signal, and a second gain to generate a low frequency passthrough signal. The processed low frequency signals are provided to the micro-loudspeaker by combination with one of a left channel or a right channel of the processed high frequency signal.

Abstract: An impulse response is computed between i) an audio signal that is being output as sound by a loudspeaker that is integrated in a loudspeaker enclosure, and ii) a microphone signal from a microphone that is recording the output by the loudspeaker and that is also integrated in the loudspeaker enclosure. A reverberation spectrum is extracted from the impulse response. Sound power spectrum at the listening distance is computed, based on the reverberation spectrum, and an equalization filter is determined based on i) the estimated sound power spectrum and ii) a desired frequency response at the listening distance. Other aspects are also described and claimed.

Abstract: Technologies for time-delayed augmented reality (AR) presentations includes determining a location of a plurality of user AR systems located within the presentation site and determining a time delay of an AR sensory stimulus event of an AR presentation to be presented in the presentation site for each user AR system based on the location of the corresponding user AR system within the presentation site. The AR sensory stimulus event is presented to each user AR system based on the determined time delay associated with the corresponding user AR system. Each user AR system generates the AR sensory stimulus event based on a timing parameter that defines the time delay for the corresponding user AR system such that the generation of the AR sensory stimulus event is time-delayed based on the location of the user AR system within the presentation site.

Abstract: The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

Abstract: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: An apparatus comprising: an input configured to receive at least two audio signals, the at least two audio signals having a relative displacement between them; an orientation determiner configured to determine an audio capture orientation based on a device orientation and information concerning microphone configuration in the apparatus; and an audio output generator configured to output the at least two audio signals based on the capture orientation, such that playback of the at least two audio signals is performed based on the capture orientation.

Abstract: Example systems and methods represent audio using a sequence of two-dimensional (2D) Fourier transforms (2DFTs), and such a sequence may be used by a specially configured machine to perform audio identification, such as for automated cover song identification. Such systems and methods are robust to timbral changes, time skews, and pitch skews that occur in cover songs found in content repositories. The systems and methods allow copyright holders to search the content repositories for unlicensed cover song.

Abstract: An MTM loudspeaker in which an array of smaller drivers replaces each of the larger drivers in a traditional MTM design. Doing so reduces the total moving mass of the drivers reproducing the lower-frequency sounds in the MTM design, while still allowing enough air to be moved for the loudspeaker to reproduce sounds in lower frequency ranges that would normally be produced by the larger drivers and could not be adequately produced by a single instance of the smaller driver. For example, a single tweeter or supertweeter, or a pair of tweeters or supertweeters still serves a traditional tweeter or tweeter function in the MTM design (handling high frequencies), but a plurality of tweeters or supertweeters arranged in two arrays placed on either side of the tweeter(s) or supertweeter(s) handling high frequencies is tasked with reproducing lower frequencies not traditionally handled in loudspeakers by tweeters or supertweeters.

Abstract: A method for compensating for a change in back electric motive force in a transducer in a device, wherein the transducer is driven by an output signal output by an amplifier, and a back volume of the transducer is formed by an enclosure within the device. The method comprises: receiving an input signal; determining, based on a characteristic of the transducer that the transducer has moved from an equilibrium position within the device; and based on the determination, adjusting a gain applied to the input signal by the amplifier to generate the output signal.

Abstract: An effective and simple psychoacoustic bass generation system generates a harmonic signal having inter-modulation controllable to remain below a threshold level and includes a high-pass filter configured to pass harmonics which are reproducible with fidelity by the loudspeaker or other transducer and a loudness matching block configured to compensate the loudness of the desired harmonics to match the loudness of the original signal.

Abstract: A system for applying a filtering scheme to an audio signal in which one or more predefined audio filtering schemes may be applied. A user interface is provided for receiving input from a user and displaying status information to the user, and may include one of a variety of adjustment controls such as a physical knob, switch, dial, slide, button, digital touchscreen, or remote control, among others. Varying linear and nonlinear filtering schemes may be provided to impact various frequency ranges in the audio signal to generate an output signal.

Abstract: Example systems and methods represent audio using a sequence of two-dimensional (2D) Fourier transforms (2DFTs), and such a sequence may be used by a specially configured machine to perform audio identification, such as for cover song identification. Such systems and methods are robust to timbral changes, time skews, and pitch skews. In particular, a special data structure provides a time-series representation of audio, and this time-series representation is robust to key changes, timbral changes, and small local tempo deviations. Accordingly, the systems and methods described herein analyze cross-similarity between these time-series representations. In some example embodiments, such systems and methods extract features from an audio fingerprint and calculate a distance measure that is robust and invariant to changes in musical structure.