Abstract: An example reverberation device configured for use with a musical instrument is disclosed. The example reverberation device comprises: (a) an input amplifier configured to receive an input signal from the musical instrument and output an amplified input signal based on the received input signal; (b) an input transducer connected to the input amplifier and configured to transmit the amplified input signal; (c) a reverberation plate comprising a bended surface, wherein the reverberation plate is connected to the input transducer, and wherein the reverberation plate is configured to output the input signal as an output signal at the bended surface; (d) an output transducer connected to the reverberation plate at the bended surface and configured to transmit the output signal; and (e) an output amplifier configured to receive the transmitted output signal from the output transducer and output an amplified output signal.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment.

Abstract: An analysis filter bank corresponding to a plurality of sub-bands, comprising: multiple sub-filters with different center frequencies which perform multiple complex-type first-order infinite impulse response filtering operations on an audio input signal to generate multiple sub-filter signals; a first set of binomial combiners, each of which performs a weighted-sum operation on a first number of the sub-filter signals with a first set of binomial weights to generate one of multiple sub-band signals; a second set of binomial combiners, each of which performs a weighted-sum operation on a second number of the sub-filter signals with a second set of binomial weights to generate one of multiple lower sub-band-edge signals or one of multiple higher sub-band-edge signals; and multiple envelope detection with decimation devices, which perform multiple envelope detection with decimation operations on the sub-band signals, the lower sub-band-edge signals, and the higher sub-band-edge signals to generate multiple fine

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.

Abstract: Disclosed are techniques and devices which include a memory configured to store audio data within a first audio zone, or a second audio zone in a layered soundfield. The memory is coupled to one or more processors and the memory is configured to store the audio data in the first audio zone and the second audio data in the layered soundfield. The one or more processors are configured to receive an interaction command to control the audio data in the first audio zone and the second audio zone in the layered soundfield, and generate one or more indicators that the interaction command was received to control the audio data, in the first audio zone or the second audio zone of the layered soundfield.

Abstract: An illustrative method for estimating a direct-to-reverberant ratio of a sound signal is described, wherein the direct-to-reverberant ratio is indicative of a ratio between direct sound received from a sound source and reverberated sound received from reflections in an environment of the sound source. The method includes determining a first energy value of a sound signal for a first time frame; assigning to an onset value of the first time frame a positive value, if the difference of the first energy value of the first time frame and a second energy value of a preceding second time frame is greater than a threshold, and a zero value otherwise; and determining the direct-to-reverberant ratio by providing an onset signal comprising the onset value to a machine learning algorithm, which has been trained to determine the direct-to-reverberant ratio based on the onset signal.

Abstract: In one implementation, a method of transforming a sound into a virtual sound for a synthesized reality (SR) setting is performed by a head-mounted device (HMD) including one or more processors, non-transitory memory, a microphone, a speaker, and a display. The method includes displaying, on the display, an image representation of a synthesized reality (SR) setting including a plurality of surfaces associated with an acoustic reverberation property of the SR setting. The method includes recording, via the microphone, a real sound produced in a physical setting. The method further includes generating, using the one or more processors, a virtual sound by transforming the real sound based on the acoustic reverberation property of the SR setting. The method further includes playing, via the speaker, the virtual sound.

Abstract: Examples of the disclosure describe systems and methods for estimating acoustic properties of an environment. In an example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of a wearable head device, wherein the second audio signal is based on the second reverberation time.

Abstract: A signal processing device includes a power estimating unit that treats the feature quantity of a signal including reverberation as the input; inputs an observation feature quantity corresponding to an observation signal to a neural network which is learnt in such a way that the estimate value of the feature quantity corresponding to the power of the signal having reduced reverberation, from among the input signal, is output; and estimates the estimate value of the feature quantity corresponding to the power of the signal having reduced reverberation and corresponding to the observation signal. Moreover, the signal processing device includes a regression coefficient estimating unit that uses the estimate value of the feature quantity corresponding to the power as obtained as the estimation result by the power estimating unit, and estimates a regression coefficient of the autoregressive process for generating the observation signal.

Abstract: Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

Abstract: The present technology relates to an audio output controller, an audio output control method, and a program that can improve sound quality. An audio output controller includes multiple speaker units installed so as to face different directions, outputs measurement sound from at least one speaker unit of the multiple speaker units, and controls a gain of the speaker unit on the basis of a reverberation characteristic when the measurement sound is measured by a microphone in a predetermined position. Measurement sound output from a speaker unit installed in another audio output controller is measured by the microphone. Alternatively, measurement sound output from an installed speaker unit is measured by the microphone. The present technology can be applied to a wireless speakers, for example.

Abstract: A system for enhancing artificial reverberation in a noisy listening space includes a sensor configured to generate a signal indicating a current noise level in a listening space, a loudspeaker configured to output sound in the listening space based on an output signal, and at least one processor.

Abstract: Determination of material acoustic parameters for a headset is presented herein. A value of a material acoustic parameter is initialized. A simulation is performed using the value of the material acoustic parameter and a model. The model includes a three-dimensional representation of a local area occupied by the headset. During the simulation, the value of the material acoustic parameter is dynamically modified until a reverberation time calculated based on the modified value of the material acoustic parameter falls within a threshold value of a target reverberation time. The model is updated with the modified value of the material acoustic parameter. The model is used to determine one or more acoustic parameters. Audio content is rendered based on the one or more acoustic parameters so that the audio content appears originating from an object in the local area.

Abstract: The disclosure describes one or more embodiments of an impulse response system that generates accurate and realistic synthetic impulse responses. For example, given an acoustic impulse response, the impulse response system can generate one or more synthetic impulse responses that modify the direct-to-reverberant ratio (DRR) of the acoustic impulse response. As another example, the impulse response system can generate one or more synthetic impulse responses that modify the reverberation time (e.g., T60) of the acoustic impulse response. Further, utilizing the synthetic impulse responses, the impulse response system can perform a variety of functions to improve a digital audio recording or acoustic measurement or prediction model.

Abstract: A method, computer program product, and computer system for measuring, by a computing device, a plurality of Room Impulse Responses (RIRs) associated with a set of two or more microphones. At least a portion of the RIRs may be augmented. At least the portion of the RIRs may be converted to their respective Relative Transfer Function (RTF) representations. The RTF representations may be applied to training data to generate an acoustic model for automatic speech recognition.

Abstract: According to the present disclosure, a sound output device includes a sound acquisition part that acquires a sound signal generated from an ambient sound, a reverb process part that performs a reverb process on the sound signal, and a sound output part that outputs a sound generated from the sound signal subjected to the reverb process, to a vicinity of an ear of a listener. This configuration allows a listener to hear sound acquired in real time to which desired reverberation is added.

Abstract: A sound processing device includes: a sound source that outputs a common sound signal to a first speaker for a vehicle exterior and a second speaker for a vehicle interior that are mounted on the vehicle; a notification sound controller that outputs a signal corresponding to a notification sound for notifying an approach of the vehicle to the first speaker based on the sound signal output from the sound source; and an acceleration sound controller that outputs a signal corresponding to an acceleration sound of the vehicle to the second speaker based on the sound signal output from the sound source.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output.

Abstract: Systems and methods for providing accurate and independent control of reverberation properties are disclosed. In some embodiments, a system may include a reverberation processing system, a direct processing system, and a combiner. The reverberation processing system can include a reverb initial power (RIP) control system and a reverberator. The RIP control system can include a reverb initial gain (RIG) and a RIP corrector. The RIG can be configured to apply a RIG value to the input signal, and the RIP corrector can be configured to apply a RIP correction factor to the signal from the RIG. The reverberator can be configured to apply reverberation effects to the signal from the RIP control system. In some embodiments, one or more values and/or correction factors can be calculated and applied such that the signal output from a component in the reverberation processing system is normalized to a predetermined value (e.g., unity (1.0)).

Abstract: A sound processing device includes: a combining processor that combines a performance sound and a source sound, based on operation information corresponding to a performance operation on an instrument. The performance sound is obtained by picking up a sound generated by the performance operation on the instrument. The source sound is obtained from a sound source.

Abstract: Utilizing an adaptive multichannel technique to mitigate reverberation present in received audio signals, prior to providing corresponding audio data to one or more additional component(s), such as automatic speech recognition (ASR) components. Implementations disclosed herein are “adaptive”, in that they utilize a filter, in the reverberation mitigation, that is online, causal and varies depending on characteristics of the input. Implementations disclosed herein are “multichannel”, in that a corresponding audio signal is received from each of multiple audio transducers (also referred to herein as “microphones”) of a client device, and the multiple audio signals (e.g., frequency domain representations thereof) are utilized in updating of the filter—and dereverberation occurs for audio data corresponding to each of the audio signals (e.g., frequency domain representations thereof) prior to the audio data being provided to ASR component(s) and/or other component(s).

Abstract: Certain example embodiments relate to speech privacy systems and/or associated methods. The techniques described herein disrupt the intelligibility of the perceived speech by, for example, superimposing onto an original speech signal a masking replica of the original speech signal in which portions of it are smeared by a time delay and/or amplitude adjustment, with the time delays and/or amplitude adjustments oscillating over time. In certain example embodiments, smearing of the original signal may be generated in frequency ranges corresponding to formants, consonant sounds, phonemes, and/or other related or non-related information-carrying building blocks of speech. Additionally, or in the alternative, annoying reverberations particular to a room or area in low frequency ranges may be “cut out” of the replica signal, without increasing or substantially increasing perceived loudness.

Abstract: Methods and systems for modifying ambient sound waves passing through a wall are disclosed. The wall has a first side and a second side with a cavity therebetween. A sound producing device is situated between the first side and the second side and is configured to produce interfering sound waves in the cavity such that the sound waves passing through the wall are reduced, masked, or reduced and masked.

Abstract: Provided is a signal processing apparatus that can replicate, in a real space, an environment different from the real space by granting an acoustic characteristic different from that of the real space, to a sound released in the real space. There is provided a signal processing apparatus including a control unit configured to decide a predetermined acoustic characteristic for causing a user to hear a collected ambient sound of the user in a space having a different acoustic characteristic, in accordance with content being reproduced, or an action of a user, and to add the decided acoustic characteristic to the ambient sound.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: In music recording and virtual reality applications, it is desired to control the perceived size of a synthesized acoustic space. In an embodiment, a room size parameter is used to modify characteristics of an artificial reverberator so as to affect a listener's sense of the size of the acoustic space. Properties of the reverberation related to perceived room size such as decay time and equalization are adjusted interactively.

Abstract: The application relates to a method of reducing reverberation in an audio processing device and to an audio processing device. The object of the present application is to provide an alternative method of reducing noise, e.g. reverberation, in a sound signal. The method comprises the steps of a) providing a time variant electric input signal representative of a sound; b) providing a logarithmic representation of said electric input signal; c) providing a predefined statistical model of the likelihood that a specific slope of the logarithmic representation of the electric input signal is due to reverberation; d) identifying time instances of the electric input signal being reverberant according to the statistical model; and e) applying an attenuation to the time instances identified as reverberant. This has the advantage of providing an enhanced sound signal. The invention may e.g. be used for enhancing noisy, e.g. reverberant, signals.

Abstract: The present technology relates to an audio processing device, a method therefor, and a program therefor capable of achieving more flexible audio reproduction. An input unit receives input of an assumed listening position of sound of an object, which is a sound source, and outputs assumed listening position information indicating the assumed listening position. A position information correction unit corrects position information of each object on the basis of the assumed listening position information to obtain corrected position information. A gain/frequency characteristic correction unit performs gain correction and frequency characteristic correction on a waveform signal of an object on the basis of the position information and the corrected position information.

Abstract: A method of processing a series of microphone inputs of an audio conference, the method including the steps of: (a) conducting a spatial analysis and feature extraction of the audio conference based on current audio activity; (b) aggregating historical information to obtain information about the approximate relative location of recent sound objects relative to the series of microphone inputs; (c) utilizing the relative location or distance of the sound objects from the series of microphone inputs to determine if beam forming should be utilized to enhance the audio reception from recent sound objects.

Abstract: An audio object including audio content and object metadata is received. The object metadata indicates an object spatial position of the audio object to be rendered by audio speakers in a playback environment. Based on the object spatial position and source spatial positions of the audio speakers, initial gain values for the audio speakers are determined. The initial gain values can be used to select a set of audio speakers from among the audio speakers. Based on the object spatial position and a set of source spatial positions at which the set of audio speakers are respectively located in the playback environment, a set of non-negative optimized gain values for the set of audio speakers is determined. The audio object at the object spatial position is rendered with the set of optimized gain values for the set of audio speakers.

Abstract: The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well.

Abstract: A signal processing apparatus has a first memory in which plural pieces of FIR coefficient data used for implementing an FIR filter algorithm are stored, a second memory which stores plural pieces of input data to be subjected to the FIR filter algorithm, and a processor implements the FIR filter algorithm using the plural pieces of FIR coefficient data stored in the first memory and the plural pieces of input data stored in the second memory as many times as the number corresponding to a designated filter order, in which filter algorithm each piece of coefficient data and each piece of input data are multiplied together and resultant products are summed up. The signal processing apparatus is provided, which can implement plural sorts of FIR filter algorithms of filter order which can be changed flexibly.

Abstract: A method for estimating the acoustic reverberations in an environment comprising the following steps: a measurement step in which one acoustic signal emitted in the environment is captured; a step for determination of acoustic energy decay rate distribution during which an acoustic energy decay rate distribution is determined from the acoustic signal captured in step (a); an estimation step during which a reverberation time and a reverberation level of sound in the environment are estimated by regression from the characteristic function of the acoustic energy decay rate distribution determined in step (b).

Abstract: Holographic sound is recorded and reproduced by way of a single monaural recording per left and right ear recorded. This is accomplished by determining the phase shift of frequencies recorded after dividing the sound into discrete frequencies in a recording device having resonators, each resonating at a different frequency, placed in a circular arrangement and divided into discrete channels by non-resonant material. The resonators are placed in a pseudo-randomized arrangement within the recording device and the circle of resonators is in front of a microphone which records the sound monaurally. Playback is then by way of arranging speakers or transducers into micro perforated sheets which amplify the sound, the arrangement of speakers/transducers around a central point. The sound is then played back directionally based on the position where the sound originally was recorded from and the position of the particular transducer around the central point.

Abstract: Systems and methods are provided for conducting conference calls using doppler-based, i.e., reverberation-based techniques. The embodiments comprise a call device performing operations to join a call session hosted on a session server; receive sensor data comprising an audio signal from a first microphone and location information associated with the first microphone; determine a reverberation parameter associated with the location information; generate a first processed audio signal based on the audio signal and the reverberation parameter; and transmit the first processed audio signal to the session server. The session server may perform operations to receive a respective processed audio signal; determine a sound quality parameter of the respective processed audio signal; generate a balanced audio signal based on the sound quality parameter and the received processed audio signal; and transmit the balanced audio signal to a remote call device belonging to a second party.

Abstract: In embodiments of the present invention improved capabilities are described for a computer-based method for reproducing a performance sound quality in a rehearsal space, the method comprising accessing a computer stored multi-dimensional sound signature for an audience location in a performance space, receiving sound data from a sound input device in the rehearsal space, modifying the sound data to match a sound characteristic of the multi-dimensional sound signature, and transmitting the modified sound data through a sound output device in the rehearsal space.

Abstract: Systems and methods are provided for conducting conference calls using doppler-based, i.e., reverberation-based techniques. The embodiments comprise a call device performing operations to join a call session hosted on a session server; receive sensor data comprising an audio signal from a first microphone and location information associated with the first microphone; determine a reverberation parameter associated with the location information; generate a first processed audio signal based on the audio signal and the reverberation parameter; and transmit the first processed audio signal to the session server. The session server may perform operations to receive a respective processed audio signal; determine a sound quality parameter of the respective processed audio signal; generate a balanced audio signal based on the sound quality parameter and the received processed audio signal; and transmit the balanced audio signal to a remote call device belonging to a second party.

Abstract: Provided are a sound effect processing method and device. The sound effect processing method includes: invoking a pre-loaded plug-in unit manager to acquire a sound effect processing parameter supported by each preloaded sound effect plug-in unit; acquiring a sound effect configuration file pre-configured by the plug-in unit manager; displaying sound effect index identifiers corresponding to various sound effect modes; according to a selected sound effect index identifier, determining a selected sound effect mode, and according to an adjustment interface, acquiring adjusted parameter control data; sending the adjusted parameter control data to a selected sound effect plug-in unit; and invoking the plug-in unit manager to send data to be processed to the selected sound effect plug-in unit, and according to the adjusted parameter control data, conducting sound effect processing on the data to be processed by the selected sound effect plug-in unit.

Abstract: A system, article, and method of acoustic dereverberation factoring the actual non-ideal acoustic environment.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: An effect providing apparatus is provided which ostensibly increases the number of effects that can be simultaneously provided. When switching to flanger processing is performed while filter processing in accordance with a periodic signal (LFO signal) is being performed in an effect processing section that can provide only one effect, a CPU advances the periodic signal (LFO signal) from a phase at the time of the switching, and performs the filter processing in accordance with the advancing periodic signal when the flanger processing is ended. As a result of this configuration, the number of effects that can be simultaneously provided is ostensibly increased.

Abstract: A delay line is constructed by combining a phase generator and a fabric. The phase generator splits a digital input signal in multiple incrementally delayed versions, which are input to the fabric. The fabric has an array of node filters. Inputs of filters in the first array column are inputs of the fabric. A node filter has a delay element and a cross-coupling element, whose output signals are added or subtracted to form a filter output signal. A node filter in a row is concatenated to the previous filter in the row through its delay element. Inputs of cross-coupling elements are connected to other array rows. Outputs of node filters form the outputs of the fabric. Delay times of delay elements and cross-coupling elements are nominally equal. Drive strengths of cross-coupling elements may be lower than drive strengths of delay elements.

Abstract: A method and apparatus may be used to perform personalized audio virtualization. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: An information processing device includes: a reflecting surface determining section configured to determine a reflecting surface as an object for reflecting a sound; a reflecting surface information obtaining section configured to obtain reflecting surface information indicating a reflection characteristic of the determined reflecting surface; and an output control portion configured to output a directional sound according to the obtained reflecting surface information to the determined reflecting surface.

Abstract: Methods and devices for processing and voice operated control are provided. The method can include performing a non-difference comparison between a first received sound and a second received sound, determining if speech exists based on the comparison, and transmitting or providing a decision that the speech is present to at least one among the device, a cell phone, a media player, or a portable computing device. Other embodiments are disclosed.

Abstract: A reverberator for reverberating an audio signal includes a feedback delay loop processor for delaying at least two different frequency subband signals representing the audio signal by different loop delays to obtain reverberated frequency subband signals.

Abstract: An effect giving device that can solve a problem that sound is localized inside a head when using a headphone and a problem that timbre changes depending on kind of the headphone is provided. A DSP 10 performs a speaker adjusting equalizer process that processes an entire bandwidth with seven bands. When output selecting means select a headphone 320, the DSP 10 performs a headphone adjusting equalizer process of the entire bandwidth with three bands of the equalizer process, and processes a process of other bands by replacing it to a lateralization sense alleviating process for the headphone 320. Since the headphone 320 and a speaker 318 are exclusively used, using an identical program storage area efficiently ensures an alleviation of a lateralization sense when using the headphone 320 and an adjustment of the timbre.

Abstract: A sound effect producing apparatus includes an input portion, a memory portion, a pseudo sound reflection producing portion, and an effect provision portion. The input portion performs inputting of an audio signal. The memory portion stores sound effect information that includes production information for producing a pseudo sound reflection corresponding to a sound reflection generated in a predetermined acoustic space and sound source position information showing a sound source position of the pseudo sound reflection. The pseudo sound reflection producing portion produces a pseudo sound reflection based on the production information. The effect provision portion performs a process of localizing the pseudo sound reflection using a predetermined direction as a reference, based on the audio signal and the sound source position information.

Abstract: A mixing signal processing technique modifies digital audio recordings to simulate the linear and nonlinear effects of propagation and mixing of sounds in air. When multiple sounds or complex sounds comprised of multiple frequencies in the audible spectrum propagate in such a nonlinear medium, they transfer energy into sound at new frequencies given by the sums and differences of the original signal frequencies. The mixing signal processing technique may improve the ability of a system to reproduce the effects of a live performance using a digital audio recording.

Abstract: Example techniques may involve multiple calibrations for one or more playback devices. An example implementation may involve detecting, via a microphone, calibration sounds as emitted by one or more playback devices during a calibration sequence, perhaps by recording first samples while the microphone is in motion through a given environment and recording second samples while the microphone is stationary at one or more particular locations. The implementation may also include determining a first calibration for the one or more playback devices based on at least the first samples of the calibrations sounds and determining a second calibration for the one or more playback devices based on at least the second samples of the calibrations sounds. The implementation may further include applying at least one of (a) the first calibration or (b) the second calibration to playback by the one or more playback devices.