Abstract: The present embodiments relate to audio effect processing, and more particularly to a method for creating reverberation impulse responses from prerecorded or live source materials forms the basis of a family of reverberation effects. In one embodiment, segments of audio are selected and processed to form an evolving sequence of reverberation impulse responses that are applied to the original source material—that is, an audio stream reverberating itself. In another embodiment, impulse responses derived from one audio track are applied to another audio track. In a further embodiment, reverberation impulse responses are formed by summing randomly selected segments of the source audio, and imposing reverberation characteristics, including reverberation time, wet equalization, wet-dry mix, and predelay. By controlling the number and timing of the selected source audio segments, the method produces a collection of impulse responses that represent a trajectory through the source material.

Abstract: A method performed by a programmed processor of an electronic device. The device obtains an audio signal, obtains, using one or more microphones, a microphone signal that includes audio of an environment in which the electronic device is located. The device determines a context of the device, and selects a volume compensation model from several models based on the determined context. The device processes the audio signal according to the selected volume compensation model and the microphone signal, and uses the processed audio signal to drive one or more speakers of the device.

Abstract: [Problem] To provide a technology capable of comfortably enjoying audio content via multiple channels even in a noisy environment. [Solution] A wireless terminal 3 is disposed at a listening point of a multi-channel audio device 1. The multi-channel audio device 1 plays a multi-channel audio signal as audio playback signals of a plurality of channels, and outputs an audio playback signal for each channel from the corresponding speaker 2, and the wireless terminal 3 collects the environmental sound at the listening point, and transmits the sound collection signal to the multi-channel audio device 1. The multi-channel audio device 1 identifies, as a noise component, the difference between the sound collection signal received from the wireless terminal 3 and the audio playback signals of the plurality of channels output from the plurality of speakers, generates a noise canceling signal with the opposite phase to the noise component, and outputs the noise canceling signal from any speaker 2.

Abstract: A method of processing an acoustic signal is disclosed. According to one or more embodiments, a first acoustic signal is received via a first microphone. The first acoustic signal is associated with a first speech of a user of a wearable headgear unit. A first sensor input is received via a sensor, a control parameter is determined based on the sensor input. The control parameter is applied to one or more of the first acoustic signal, the wearable headgear unit, and the first microphone. Determining the control parameter comprises determining, based on the first sensor input, a relationship between the first speech and the first acoustic signal.

Abstract: An electronic device comprises a speaker; a wireless communication interface configured to communicate with an external device, wirelessly; and a controller configured to transmit a sound source to the speaker, transmit the sound source to the external device through the wireless communication interface, obtain a first audio corresponding to the sound source output through the speaker and a second audio corresponding to the sound source output through the external device, and control an EQ (Equalizer) mode of the external device based on a comparison result between the first audio and the second audio, or output a calibrated sound source to the speaker.

Abstract: Disclosed are methods to manipulate a given parametrized haptic curve in order to yield a smooth phase function for each acoustic transducer which minimizes unwanted parametric audio. Further, the impulse response of a haptic system describes the behavior of the system over time and can be convolved with a given input to simulate a response to that input. To produce a specific response, a deconvolution with the impulse response is necessary to generate an input.

Abstract: In at least one embodiment, an audio limiter system is provided. The system includes a first limiter block, a second limiter block, and a third limiter block. The first limiter block receives an incoming audio signal having a plurality of first frequencies and limits the incoming audio signal with the plurality of first frequencies based on at least a first gain threshold to generate at least a first high audio output signal. The second limiter block receives the incoming audio signal having a plurality of second frequencies and limits the incoming audio signal with the plurality of second frequencies based on at least a second gain threshold to generate at least a first low audio output signal. The third limiter block receives the first high audio output signal and/or the first low audio output signal and limits the first high audio output signal and/or the first low audio output signal.

Abstract: An acoustic control apparatus includes a processor with hardware. The processor calculates a first relational expression between acoustic filter coefficients of acoustic filters to be applied to voice signals containing information of sounds played back by two or more sound sources, based on an amplification magnification in a sound amplification control point with respect to the sounds played back by the two or more sound sources, and on transfer functions between the sound amplification control point and the two or more sound sources. The processor calculates a second relational expression between the acoustic filter coefficients, based on information of a frequency of the voice signals, and on an interval between the two or more sound sources. The processor calculates the acoustic filter coefficients based on the first relational expression and the second relational expression.

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: System and method for processing and adjusting PCM digital audio signals using specific reverberation and equalization settings that have been determined to improve various physical health parameters measurements, as determined by conducting bio-signal testing. The system entails converting the audio signal to digital format, a digital system processor that processes the 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 a listener. Accordingly, the invention improves at least one physiological function of the listener as measured using bio-sensors in the Avatar health testing bio-sensor measuring 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: An electronic device package comprises a primary microphone having a frequency response having a first resonance frequency, and a reference microphone having a frequency response including a second resonance frequency, the primary microphone and the reference microphone configured to substantially simultaneously receive a same acoustic signal to produce a transduced signal of the primary microphone and a transduced signal of the reference microphone, the second resonance frequency of the reference microphone being different than the first resonance frequency of the primary microphone, the package having dimensions that cause the primary microphone and reference microphone to be acoustically isolated from one another at the resonance frequency of the primary microphone, there being less than 3 dB of acoustic coupling between the primary microphone and reference microphone at the first resonance frequency.

Abstract: Techniques described herein may involve audio settings based on an environment. An example implementation may involve a playback device playing back first audio content and during playback of at least a portion of the first audio content, detecting, via the at least one microphone, an audio signal. At least a portion of the detected audio signal may include a reflection of the first audio content played back by the playback device. The playback device may determine an equalization setting based on at least the determined one or more reflection characteristics and apply the determined equalization setting during playback of second audio content.

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

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 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.