Abstract: A method performed a local device that is communicatively coupled with several remote devices, the method includes: receiving, from each remote device with which the local device is engaged in a communication session, an input audio stream; receiving, for each remote device, a set parameters; determining, for each input audio stream, whether the input audio stream is to be 1) rendered individually or 2) rendered as a mix of input audio streams based on the set of parameters; for each input audio stream that is determined to be rendered individually, spatial rendering the input audio stream as an individual virtual sound source that contains only that input audio stream; and for input audio streams that are determined to be rendered as the mix of input audio streams, spatial rendering the mix of input audio streams as a single virtual sound source that contains the mix of input audio streams.

Abstract: A conference terminal and an echo cancellation method for a conference are provided. In the echo cancellation method, a synthetic speech signal is received. The synthetic speech signal includes a user speech signal of a speaking party corresponding to a first conference terminal of multiple conference terminals and an audio watermark signal corresponding to the first conference terminal. One or more delay times corresponding to the audio watermark signal are detected in a received audio signal. The received audio signal is recorded through a sound receiver of a second conference terminal of the conference terminals. An echo in the received audio signal is canceled according to the delay time.

Abstract: Disclosed are methods of encoding and decoding an audio signal using side information, and an encoder and a decoder for performing the methods. The method of encoding an audio signal using side information includes identifying an input signal, the input signal being an original audio signal, extracting side information from the input signal using a learning model trained to extract side information from a feature vector of the input signal, encoding the input signal, and generating a bitstream by combining the encoded input signal and the side information.

Abstract: A device for reducing crosstalk in an audio system that has first and second pairs of loudspeakers and first and second audio sources. The device is connected to each audio source and to at least the first pair of loudspeakers. The device includes a module for acquiring first audio signals from the first source and second audio signals from the second source, a module for determining crosstalk reduction filters resulting from a loudspeaker of the second pair, a module for calculating corrective signals, by applying the reduction filters to the second audio signals, and a module for generating corrected audio signals for the first pair, obtained from the first audio signals and corrective signals. Each reduction filter is obtained from transfer functions, each representing an acoustic path between a loudspeaker and a user's ear.

Abstract: An apparatus and method of generating personalized HRTFs. The system is prepared by calculating a model for HRTFs described as the relationship between a finite example set of input data, namely anthropometric measures and demographic information for a set of individuals, and a corresponding set of output data, namely HRTFs numerically simulated using a high-resolution database of 3D scans of the same set of individuals. At the time of use, the system queries the user for their demographic information, and then from a series of images of the user, the system detects and measures various anthropometric characteristics. The system then applies the prepared model to the anthropometric and demographic data as part of generating a personalized HRTF. In this manner, the personalized HRTF can be generated with more convenience than by performing a high-resolution scan or an acoustic measurement of the user, and with less computational complexity than by numerically simulating their HRTF.

Abstract: A system comprises a sound source to output an audio signal to a plurality of speakers arranged in a plurality of locations in a vehicle. A laser device is disposed on an object representing a human head placed in a seat of the vehicle to scan the locations of the speakers. A microphone is disposed in each ear of the object. A controller routes the audio signal to the speakers one speaker at a time, receives audio signals received via the microphones, and compiles binaural acoustic data for the vehicle based on the audio signals received via the microphones. The controller receives scan data from the laser device and generates geometric data for the vehicle based on the scan data. The controller generates head-related transfer functions (HRTFs) for the object based on the binaural acoustic data and the geometric data for the vehicle.

Abstract: An encoding system encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: A method (900) for rendering audio in a virtual reality rendering environment (180) is described. The method (900) comprises rendering (901) an origin audio signal of an origin audio source (113) of an origin audio scene (111) from an origin source position on a sphere (114) around a listening position (201) of a listener (181). Furthermore, the method (900) comprises determining (902) that the listener (181) moves from the listening position (201) within the origin audio scene (111) to a listening position (202) within a different destination audio scene (112). In addition, the method (900) comprises applying (903) a fade-out gain to the origin audio signal to determine a modified origin audio signal, and rendering (903) the modified origin audio signal of the origin audio source (113) from the origin source position on the sphere (114) around the listening position (201, 202).

Abstract: A method for supporting a multi-conversation mode for a vehicle of the present disclosure may include: receiving user information of the multi-conversation mode and at least one of a message or speech from a conversation partner participating in the multi-conversation mode; allocating sound spaces on the basis of the user information; and assigning directivity to speech generated on the basis of at least one of the message or the speech and outputting the speech to an allocated space.

Abstract: A method including: obtaining at least one spatial audio signal including at least one audio signal, wherein the at least one spatial audio signal at least partially defines an audio scene; obtaining at least one augmentation audio signal; determining at least two audio objects based upon the at least one augmentation audio signal; determining audio-object dependency information for the determined at least two audio objects; and augmenting the audio scene based, at least partially, on both the determined at least two audio objects and the determined audio-object dependency information.

Abstract: A gaming unit and producing a game-audio output of a varying volume level and a chat-voice output is connected to headphones adapted to be worn by a player of the unit and connected to the gaming unit for making the chat-voice output and the game-audio output audible to the player. A circuit or software associated with the headphones can increase the chat-voice volume level generally proportionately to the game-audio volume level.

Abstract: A feedback system may play back, to a user, an altered version of the user's voice in real time, in order to reduce stuttering by the user. The system may operate in different feedback modes at different times. For instance, the system may detect when the severity of a user's stuttering increases, which is indicative of the user habituating to the current feedback mode. The system may then switch to a different feedback mode. In some cases, the feedback modes include at least a Whisper mode, a Reverb mode, and a Harmony mode. In Whisper mode, the user's voice may be transformed to sound as if it were whispering in the user's ears. In Harmony mode, the user's voice may be altered as if the user were harmonizing with himself or herself. In Reverb mode, the user's voice may be altered so that it reverberates.

Abstract: In a reliable multi-cast, a concealment scheme may be applied to recover or conceal lost or otherwise corrupted packets of audio information for one channel based on the audio information of other channels in the reliable multi-cast. The concealment scheme may employ correction factors for channels derived from the channel relationships.

Abstract: A method for securing a computer system comprising a step of: disguising a password entry screen.

Abstract: Provided is an information processing apparatus that carries out a process of deriving a head related transfer function. The information processing apparatus includes a detection unit that detects a position of a head of a user, a storage unit that stores a head related transfer function of the user, a determination unit that determines a position of a sound source for measuring the head related transfer function of the user based on the position of the head detected by the detection unit and information stored in the storage unit, a control unit that controls the sound source to output measurement signal sound from the position determined by the determination unit, and a calculation unit that calculates the head related transfer function of the user based on collected sound data obtained by collecting, at the position of the head, the measurement signal sound output from the sound source.

Abstract: An apparatus is disclosed, configured to receive, from first and second spatial audio capture apparatuses, respective first and second composite audio signals comprising components derived from one or more sound sources in a capture space. The apparatus is further configured to identify a position of a user device corresponding to one of first and second areas respectively associated with the positions of the first and second spatial audio capture apparatuses, and to render audio representing the one or more sound sources to the user device, the rendering being performed differently dependent on, for the spatial audio capture apparatus associated with the identified first or second area, whether or not individual audio signals from each of the one or more sound sources can be successfully separated from its composite signal.

Abstract: The present disclosure provides an audio data processing circuit and an audio data processing method. The audio data processing circuit includes a word select interface, a clock signal interface and an audio data interface. The word select interface is configured to receive a word select signal. The clock signal interface is configured to receive a clock signal, and generating an audio data interface signal according to a number of clocks of the clock signal in one period of the word select signal. The audio data interface is configured to transmit the audio data to a processing unit through a first transmission protocol or a second transmission protocol.

Abstract: An apparatus for optimizing a binaural beat according to an embodiment of the present invention includes: a binaural beat generator that generates a binaural beat; and a binaural beat controller that optimizes the binaural beat in consideration of a hearing state of left and right ears of a listener. A method for optimizing a binaural beat according to an embodiment of the present invention includes: generating a binaural beat; and optimizing the binaural beat in consideration of a hearing state of left and right ears of a listener.

Abstract: A personalized sound management system for an acoustic space includes at least one transducer, a data communication system, one or more processors operatively coupled to the data communication system and the at least one transducer, and a medium coupled to the one or more processors. The processors access a database of sonic signatures and display a plurality of personalized sound management applications that perform at least one or more tasks among identifying a sonic signature, calculating a sound pressure level, storing metadata related to a sonic signature, monitoring sound pressure level dosage levels, switching to an ear canal microphone in a noisy environment, recording a user's voice, storing the user's voice in a memory of an earpiece device, or storing the user's voice in a memory of a server system, or converting received text received in texts or emails to voice using text to speech conversion. Other embodiments are disclosed.

Abstract: An example device for processing one or more audio streams includes a memory configured to store the one or more audio streams and one or more processors implemented in circuitry coupled to the memory. The one or more processors are configured to determine a listener position. The one or more processors are also configured to determine one or more clusters of the one or more audio streams. The one or more processors are also configured to determine a rendering mode based on the listener position and the one or more clusters. The device also includes a renderer configured to render at least one of the one or more clusters of audio streams based on the rendering mode.

Abstract: Environmental sound is recorded using one or more microphones. A source of the recorded environmental sound is classified. The recorded environmental sound is weighted based on the classification of the source and the source media sound using a weighting mode to determine whether to mix the recorded environmental. The recorded environmental sound is mixed with source media sound to produce a mixed sound based on the determination. The mixed sound is played over one or more speakers.

Abstract: A computer program product having a non-transitory computer-readable medium including computer program logic encoded thereon that, when performed on a surround sound audio system that is configured to render left front, right front, and center front audio signals, and also render left and right near-field binaurally-encoded audio signals, causes the surround sound audio system to develop the left and right near-field binaurally-encoded audio signals, and provide the left near-field binaurally-encoded audio signal to a left non-occluding near-field driver and provide the right near-field binaurally-encoded audio signal to a right non-occluding near-field driver.

Abstract: A content sharing system and method are disclosed. According to certain embodiments, the system may be used in a vehicle. The system may include a user interface and a controller coupled with the user interface. The controller may be configured to display a map of the vehicle on the user interface. The map may show at least a plurality of user devices in the vehicle, and an icon representing content presented by a first user device. The controller may also be configured to detect a user operation moving the icon on the user interface. The controller may further be configured to control, based on the user operation, sharing of the content between the first user device and at least a second user device.

Abstract: A method and apparatus for performing binaural rendering of an audio signal are provided. The method includes identifying an input signal that is based on an object, and metadata that includes distance information indicating a distance to the object, generating a binaural filter that is based on the metadata, using a binaural room impulse response, obtaining a binaural filter to which a low-pass filter (LPF) is applied, using a frequency response control that is based on the distance information, and generating a binaural-rendered output signal by performing a convolution of the input signal and the binaural filter to which the LPF is applied.

Abstract: Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.

Abstract: Synthesizing engine sound in a cabin of a vehicle includes generating a synthesized engine noise audio signal including a plurality of engine order outputs; mixing the engine order outputs in accordance with a first set of gain levels to provide a first channel output; mixing the engine order outputs in accordance with a second set of gain levels to provide a second channel output; using a first localization filter set to provide a first localized output to localize the first channel output as a first sound image at a first location within the cabin, and a second localization filter set to provide a second localized output to localize the second channel output as a second sound image at a second location within the cabin; and mixing the first and second localized outputs into loudspeaker outputs to be applied to the loudspeakers in the cabin.

Abstract: The disclosure describes systems and methods for processing audio signals in a vehicle to perform sound source separation. The sound source separation is performed using transfer functions and involves separation of the speech of multiple occupants. The separated speech can be used to isolate and correctly respond to a command to control vehicle systems.

Abstract: User-input obfuscating techniques involving displaying multiple targets on a display device. A visual indication of the current user target moves to a first target at a first time in response to user input, and a first presentation of a visual or audio feedback signal confirming actuation of a target occurs at a second time, without a user input for actuation of a target received while the current user target has remained at the first target from the first time to the second time. The visual indication moves to a second target at a third time in response to user input. The second target is actuated in response to receiving user input for actuation of the current user target after the third time. Then, with the current user target having remained at the second target since the third time, a second presentation of the visual or audio feedback signal is provided.

Abstract: The invention provides a method for generating output filters to a plurality of loudspeakers at respective positions for playback of a plurality of different input signals in respective spatially different sound zones by means of a processor system. The method comprising computing spatio-temporal correlation matrices in response to spatial information, e.g. measured transfer functions, and in response to desired sound pressures in the plurality of sound zones. Joint eigenvalue decomposition of the spatial correlation matrices are then computed, or at least an approximation thereof, to arrive at eigenvectors accordingly. Next, variable span filters a reformed from a linear combination of the eigenvectors in response to a desired trade-off between acoustic contrast and acoustic errors in the sound zones. Finally, output filter for each of the plurality of loudspeakers, for each of the plurality of input signals, in accordance with the variable span filters.

Abstract: An audio processing apparatus comprises a receiver (705) which receives audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers (703). A renderer (707) generating audio transducer signals for the set of audio transducers from the audio data. The renderer (7010) is capable of rendering audio components in accordance with a plurality of rendering modes. A render controller (709) selects the rendering modes for the renderer (707) from the plurality of rendering modes based on the audio transducer position data. The renderer (707) can employ different rendering modes for different subsets of the set of audio transducers the render controller (709) can independently select rendering modes for each of the different subsets of the set of audio transducers (703).

Abstract: A method includes receiving a multi-channel audio signal (101) including multiple input audio channels (102, 104, 106, 108) that are configured to play audio from multiple respective locations relative to a listener. One or more spectral components that undergo a panning effect (1001, 1002, 1003) are identified in the multi-channel audio signal among at least some of the input audio channels. One or more virtual channels (1100, 1200, 1300) are generated, which together with the input audio channels form an extended set (111) of audio channels that retain the identified panning effect. A reduced set (222) of output audio signals, fewer in number than the input audio signals, is generated from the extended set, including recreating the panning effect in the output audio signals. The reduced set of output audio signals is outputted to a user.

Abstract: In one implementation, a method of changing an audio property of an object is performed at a device including one or more processors coupled to non-transitory memory. The method includes displaying, using a display, a representation of a scene including a representation of an object associated with an audio property. The method includes displaying, using the display, in association with the representation of the object, a manipulator indicating a value of the audio property. The method includes receiving, using one or more input devices, a user input interacting with the manipulator. The method includes, in response to receiving the user input, changing the value of the audio property based on the user input and displaying, using the display, the manipulator indicating the changed value of the audio property.

Abstract: An audio processing device, including: a trans-aural processing unit configured to perform trans-aural processing with respect to a predetermined audio signal; and a correction processing unit configured to perform correction processing in accordance with a change in a listening position with respect to the audio signal having been subjected to the trans-aural processing.

Abstract: A system and method of adjusting an audio output in one location so that its propagation into another location is reduced. As a first device in a first location generates sound, a second device in a second location detects the propagated sound. The first device then adjusts its output based on the detected sound.

Abstract: An encoding system encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: A sound adjustment method is disclosed. The sound adjustment method includes the following the steps: receiving a sound adjustment command; receiving a right channel sound signal and a left channel sound signal; selecting a corresponding adjustment mode according to the sound adjustment command and processing the right channel sound signal and the left channel sound signal based on the adjustment mode to generate a right channel first sound signal and a left channel first sound signal, wherein different adjustment modes have different intensity adjustment levels; shifting the frequency of the right channel first sound signal by X Hz to generate a right channel second sound signal and shifting the frequency of the left channel first sound signal by Y Hz to generate a left channel second sound signal, wherein 0.5?|X?Y|?100; outputting the right channel second sound signal and the left channel second sound signal.

Abstract: Provided are a method and apparatus for localizing a multichannel sound signal. The method includes: obtaining a multichannel sound signal to which sense of elevation is applied by applying a first filter to an input sound signal; determining at least one frequency range of a dynamic cue according to change of a head-related transfer function (HRTF) indicating information regarding paths from a spatial location of an actual speaker to ears of an audience; and applying a second filter to at least one sound signal, corresponding to the determined at least one frequency range, of at least one channel in the multichannel sound signal to change the at least one sound signal so as to remove or to reduce the dynamic cue when the multichannel sound signal is output.

Abstract: A 360 degree interactive studio can include an arrangement of screens that enclose a performance area. The screens can be used to display an audience around the performance area and can provide an opening through which performers can enter the performance area. A 360 degree arrangement of speakers can be used to project audio of the displayed audience into the presentation area. A 360 degree arrangement of lighting can also be used to illuminate the performance area. In this way, the 360 degree interactive studio can create a live event environment for the performers and the audience even though the audience is remotely located.

Abstract: A vehicle is provided to minimize the difference in left and right output intensity of a headrest speaker by adjusting an output intensity of based on the position of the passenger's ear. The headrest includes a left-side speaker provided on the left-side of the headrest of a seat inside the vehicle to output sound and a right-side speaker provided on the right-side of the headrest to output sound. A pressure sensor measures a force applied to the headrest and a controller adjusts the output intensity of the left speaker and the right speaker based on an action point of the force measured by the pressure sensor.

Abstract: A method (900) for rendering audio in a virtual reality rendering environment (180) is described. The method (900) comprises rendering (901) an origin audio signal of an origin audio source (113) of an origin audio scene (111) from an origin source position on a sphere (114) around a listening position (201) of a listener (181). Furthermore, the method (900) comprises determining (902) that the listener (181) moves from the listening position (201) within the origin audio scene (111) to a listening position (202) within a different destination audio scene (112). In addition, the method (900) comprises applying (903) a fade-out gain to the origin audio signal to determine a modified origin audio signal, and rendering (903) the modified origin audio signal of the origin audio source (113) from the origin source position on the sphere (114) around the listening position (201, 202).

Abstract: A digital audio array circuit is provided. The digital audio array circuit includes at least two digital audio units and a system master unit. Each of the digital audio units is configured to transform a received sound wave to a digital audio signal. Each of the digital audio units includes a left/right channel configuration input terminal. The system master unit is connected to the at least two digital audio units in time division multiplexing to receive the digital audio signals. The left/right channel configuration input terminal of each of the digital audio units is configured to receive a same synchronizing signal.

Abstract: A gaming unit and producing a game-audio output of a varying volume level and a chat-voice output is connected to headphones adapted to be worn by a player of the unit and connected to the gaming unit for making the chat-voice output and the game-audio output audible to the player. A circuit or software associated with the headphones can increase the chat-voice volume level generally proportionately to the game-audio volume level.

Abstract: A method for increasing stability of an inter-channel time difference (ICTD) parameter in parametric audio coding, wherein a multi-channel audio input signal comprising at least two channels is received. The method comprises obtaining an ICTD estimate, ICTDest(m), for an audio frame m and a stability estimate of said ICTD estimate, and determining whether the obtained ICTD estimate, ICTDest(m), is valid. If the ICTDest(m) is not found valid, and a determined sufficient number of valid ICTD estimates have been found in preceding frames, a hang-over time is determined using the stability estimate and a previously obtained valid ICTD parameter, ICTD(m?1), is selected as an output parameter, ICTD(m), during the hang-over time. The output parameter, ICTD(m), is set to zero if valid ICTDest(m) is not found during the hang-over time.

Abstract: An apparatus including circuitry configured for: obtaining at least one spatial audio signal including at least one audio signal, wherein the at least one spatial audio signal defines an audio scene forming at least in part media content; rendering an audio scene based on the at least one spatial audio signal; obtaining at least one augmentation audio signal; transforming the at least one augmentation audio signal to at least two audio objects; augmenting the audio scene based on the at least two audio objects.

Abstract: A speaker excursion characterizing system for a speaker includes a signal generator configured to generate a signal at a plurality of different amplitudes. The signal may be generated at a plurality of frequencies. An inverse excursion filter has an inverse excursion filter response, receives the signal and applies the inverse excursion filter response and has an output in communication with an amplifier circuit of the speaker. The inverse excursion filter response is an inverse of an excursion filter response of an excursion filter in the amplifier circuit of the speaker.

Abstract: A device for processing audio signals, including: first and second input units providing first and second input signals of first and second audio tracks, a decomposition unit to decompose the first input audio signal to obtain a plurality of decomposed signals, a playback unit configured to start playback of a first output signal obtained from recombining at least a first decomposed signal at a first volume level with a second decomposed signal at a second volume level, such that the first output signal substantially equals the first input signal, and a transition unit for performing a transition between playback of the first output signal and playback of a second output signal obtained from the second input signal. The transition unit has a volume control section adapted for reducing the first and second volume levels according to first and second transition functions.

Abstract: Embodiments of this application provide an audio playback circuit and a terminal. The audio playback circuit includes: a first current path, configured to shunt a current output by a left-channel circuit to adjust a second voltage fed back to a first end of a right-channel headset when the left-channel circuit outputs a left-channel audio signal, where when the left-channel circuit outputs the left-channel audio signal and a right-channel circuit does not output a right-channel audio signal, the second voltage is equal to a voltage at a second end of a right-channel headset; and a second current path, configured to shunt a current output by the right-channel circuit to adjust a first voltage fed back to a first end of a left-channel headset when the right-channel circuit outputs the right-channel audio signal. The audio playback circuit in the embodiments of this application can improve isolation between a left channel and a right channel.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: An acoustic device includes: an imaging device configured to take a sample image of a space as a sound field and create an image data on the space based on the taken sample image; a sound collector configured to collect a sound generated in the space or to collect a previously-collected acoustic data therein; and a computation part configured to previously compute a plurality of parameters relevant to a coefficient of spatial acoustic filter corresponding to the sample image of the space and previously learn a sound field model of the space shown in the sample image. The computation part is configured to construct a sound field model of the sample image taken by the imaging device or of a previously-taken sample image, from the acoustic data collected by the sound collector, using the coefficient of spatial acoustic filter.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.