Abstract: In some embodiments, virtualization methods for generating a binaural signal in response to channels of a multi-channel audio signal, which apply a binaural room impulse response (BRIR) to each channel including by using at least one feedback delay network (FDN) to apply a common late reverberation to a downmix of the channels. In some embodiments, input signal channels are processed in a first processing path to apply to each channel a direct response and early reflection portion of a single-channel BRIR for the channel, and the downmix of the channels is processed in a second processing path including at least one FDN which applies the common late reverberation. Typically, the common late reverberation emulates collective macro attributes of late reverberation portions of at least some of the single-channel BRIRs. Other aspects are headphone virtualizers configured to perform any embodiment of the method.

Abstract: This invention corresponds to a device for binaural capture of sound comprising an upper module, a first external ear and a second external ear incorporated into the upper module in a removable way; two transducers, a transducer is arranged in the first external ear and the other transducer is arranged in the second external ear; and, a clamping-coupling located in the lower part of the upper module.

Abstract: A headphone arrangement includes an ear cup configured to be arranged to at least partly surround an ear of a user to define an at least partly enclosed volume about the ear of the user. The ear cup includes an at least partially hollow frame configured to at least partially enframe the ear of the user when the ear cup is arranged to surround the ear of the user. The frame includes a first cavity, the first cavity being formed by wall portions of the frame. The headphone arrangement includes at least one loudspeaker arranged within wall portions of the first cavity. The wall portions of the first cavity form a first waveguide configured to guide sound radiated from the loudspeaker through a waveguide output of the first waveguide. The waveguide output of the first waveguide includes one or more openings in the first cavity.

Abstract: A sound reproducing method used in sound reproducing apparatus that includes the steps outlined below is provided. An input sound signal related to listener data and sound source data is received. An encoding process is performed by multiplying the input sound signal by an encoding function matrix having entries related to a basis function to generate an encoding result. A decoding function matrix is retrieved from the storage and at least one direction parameter is applied to the decoding function matrix, wherein the decoding function matrix compensates a difference between an ideal approximation result and a modeled approximation result of the input sound signal. A decoding process is performed by multiplying the encoding result by the decoding function matrix having the direction parameter applied to generate an output sound signal. The output sound signal is reproduced.

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: To provide a video display system for allowing a user to look at and listen to video and sound on various means. A video display system has a video display device and a portable video display device, connected to each other for communication. The video display device produces own device video data and own device sound data to be displayed and reproduced, respectively, on the video display device; displays the own device video data and reproduces the own device sound data; encodes these data into a data format which the portable video display device is able to handle, to thereby produce other device video data and other device sound data; and sends to the portable video display device. The portable video display device receives these data, and decodes and displays the other device video data, and decodes and reproduces the other device sound data.

Abstract: Various implementations include systems and approaches for binaural testing. In one implementation, a system includes a binaural test dummy including a body having: a head-and-neck region; and a set of head-mounted microphones coupled with the head-and-neck region at anatomically correct ear locations; and a control system coupled with the binaural test dummy for incrementally modifying a position of the binaural test dummy across a range of motion.

Abstract: A method of operating an audio system that includes a wearable personal acoustic device having an acoustic driver and an auxiliary acoustic driver includes generating a first acoustic signal having a range of acoustic frequencies at the acoustic driver. A first change of operational mode of the wearable personal acoustic device is requested. In response to the request, a second acoustic signal having a first sub-range of the acoustic frequencies is generated at the acoustic driver and a third acoustic signal having a second sub-range of the acoustic frequencies is generated at the auxiliary acoustic driver. The first sub-range of the acoustic frequencies is different from the second sub-range of the acoustic frequencies and the range of acoustic frequencies is inclusive of the first and second sub-ranges of the acoustic frequencies.

Abstract: An encoding system (400) 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 (200), a decorrelating section (101) outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section (102) maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section (103) 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 (104) combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: An acoustic device that is adapted to be worn on the body of a user, with a first acoustic transducer and a second acoustic transducer, where the first transducer is closer to the expected location of a first ear of the user than is the second transducer, a third acoustic transducer and a fourth acoustic transducer, where the third transducer is closer to the expected location of a second ear of the user than is the fourth transducer, and a controller that is adapted to independently control the phase and frequency response of the first, second, third and fourth transducers.

Abstract: Systems and methods of calibrating listening devices are disclosed herein. In one embodiment, a method of calibrating a listening device (e.g., a headset) includes determining head related transfer functions (HRTF) corresponding to different parts of the user's anatomy. The resulting HRTFs are combined to form a composite HRTF.

Abstract: A three-dimensional sound reproducing method includes: acquiring a multichannel audio signal; rendering signals to a channel to be reproduced according to channel information and a frequency of the multichannel audio signal; and mixing the rendered signals.

Abstract: A method captures binaural sound of a voice of a first user with microphones located at left and rights ears of a dummy head. The dummy head transmits the voice of the first user to a portable electronic device with or near the first user. This portable electronic device transmits the binaural sound over one or more networks to another electronic device being used by a second user to communicate with the first user during the electronic call.

Abstract: An interactive in-room show and game system configured to provide engaging and immersive multimedia shows or presentations in a room. The in-room show and game system generally includes a controller for selectively operating a display device, a video projector, an audio system (e.g., one or more speakers), and other show components (e.g., one or more lights, a fan, a mechanical device, and so on). Media content is predefined or generated in real time for the room, and the controller operates (such as in response to a triggering switch or remote control device that may take the form of a show prop activated by someone in the room) to serve the media content interactively, based on user input sensed in the room by one or more sensors of a sensor assembly, to tell a story and/or magically transform the space into a multidimensional and immersive entertainment or gameplay space.

Abstract: An example implementation involves a playback device receiving digital data representing audio content, the digital data encoded in a first format. The playback device causes one or more speaker drivers to playback the audio content. The playback device decodes a portion of the received digital data to convert the portion of the received digital data from the first format to a second format and transmits, via a network interface to a computing device of an identification system, the decoded portion of the received digital data. The playback device receives, from via the network interface from the identification system, metadata corresponding to the audio content, and in response, causes a control device to display a graphical representation of the received metadata, wherein causing the control device to display the graphical representation comprises sending, via the network interface to the control device, the received metadata to the control device.

Abstract: An encoding system (400) encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters (C, P). In a decoding system (200), a decorrelating section (101) outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section (102) maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section (103) 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 (104) combines outputs from the upmix sections to obtain a reconstructed signal (X) corresponding to the signal to be reconstructed.

Abstract: The present invention is applicable to the field of audio collection and provides a fixed apparatus and an audio collection device. Multiple groups of simulated ear structures are disposed on the fixed apparatus, and the simulated ear structures have simulated ear canal openings used for placing audio collectors. The audio collectors may placed in simulated ear canal openings of the simulated ear structures of the fixed apparatus, so that collected audio signals have richer sources, higher quality, and higher stereoscopic sensation. The fixed apparatus may be used for audio recording in multiple orientations, so that smoothness, clearness, and comfort during sound field fusion and switching are ensured, more complete and richer sound field information is collected, the best sound field naturalness can be obtained, and a requirement for synchronization between a sound and an image in a VR panoramic video technology is met.

Abstract: The present invention is applicable to the field of audio recording and provides an audio collection apparatus. Multiple fixed apparatuses are stacked and have ear structures and face contours, and audio collectors are disposed at ear canal openings of the ear structures to collect audio information. During audio collection, faces of the fixed apparatuses have different horizontal orientations, so that the audio collectors obtain the audio information at the ear canal openings of the ear structures towards horizontal directions. Therefore, rich sound sources and sound authenticity are ensured, sounds are rich in quality, a stereo field perception effect is achieved, a real sound field relationship can be restored, an in-head effect is avoided, and a requirement for real synchronization between a sound and an image in a virtual reality (VR) panoramic video technology is met.

Abstract: The present invention is applicable to the field of audio recording and provides a recording device and a fixed apparatus.

Abstract: Methods and audio processing units for generating an object based audio program including conditional rendering metadata corresponding to at least one object channel of the program, where the conditional rendering metadata is indicative of at least one rendering constraint, based on playback speaker array configuration, which applies to each corresponding object channel, and methods for rendering audio content determined by such a program, including by rendering content of at least one audio channel of the program in a manner compliant with each applicable rendering constraint in response to at least some of the conditional rendering metadata. Rendering of a selected mix of content of the program may provide an immersive experience.

Abstract: Embodiments relate generally to devices and methods for improved active noise cancellation hearing protection. For example, multiple speakers might be used in each earcup, with each speaker providing active noise cancellation for a narrow frequency range which is a portion of the total frequency range of the entire array of speakers. By dividing the active noise cancellation up across an array of speakers, each speaker can better target its specific narrow frequency range, resulting in improved active noise cancellation.

Abstract: One embodiment of a headphone in which both the left and right stereo channels are contained within a single earpiece (10) and separated by vertical positioning above (12) and below (14) the ear, rather than each ear receiving one stereo channel, as in traditional headphones. This allows the listener to hear the audio content of both stereo channels with a single ear, while still being able to distinguish between the two channels. The other earpiece of the headphone may be a “dummy” (20) which contains no speaker, and the headphone can be reversed depending on which ear the user wants to listen with. Optionally, the back of each earpiece can be opened, allowing the listener to experience stereo sound in one ear while listening to his/her surroundings with the other.

Abstract: Embodiments are described for a method and system of rendering and playing back spatial audio content using a channel-based format. Spatial audio content that is played back through legacy channel-based equipment is transformed into the appropriate channel-based format resulting in the loss of certain positional information within the audio objects and positional metadata comprising the spatial audio content. To retain this information for use in spatial audio equipment even after the audio content is rendered as channel-based audio, certain metadata generated by the spatial audio processor is incorporated into the channel-based data. The channel-based audio can then be sent to a channel-based audio decoder or a spatial audio decoder. The spatial audio decoder processes the metadata to recover at least some positional information that was lost during the down-mix operation by upmixing the channel-based audio content back to the spatial audio content for optimal playback in a spatial audio environment.

Abstract: A method for performing linear mixing on coupled Head-related transfer functions (HRTFs) to determine an interpolated HRTF for any specified arrival direction in a range (e.g., a range spanning at least 60 degrees in a plane, or a full range of 360 degrees in a plane), where the coupled HRTFs have been predetermined to have properties such that linear mixing can be performed thereon (to generate interpolated HRTFs) without introducing significant comb filtering distortion. In some embodiments, the method includes steps of: in response to a signal indicative of a specified arrival direction, performing linear mixing on data indicative of coupled HRTFs of a coupled HRTF set to determine an HRTF for the specified arrival direction; and performing HRTF filtering on an audio input signal using the HRTF for the specified arrival direction.

Abstract: A boundary binaural microphone array includes a pair of microphones spaced from one another by a distance between approximately 5 cm and 30 cm. The boundary binaural microphone array has a structural support that locates the microphones no more than approximately 4 cm off of a surface upon which the array is placed. The microphones are separated by a sound barrier that provides an interaural level difference in the amplitudes of the sound signals sensed by the two microphones.

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: An audio signal enhancing device, and a corresponding method of enhancing stereophonic signals, is provided which generates an enhanced signal with improved spatial sound image quality for upmixing a stereophonic input signal. When used in combination with a centre channel processor or LFE processor, an improved processing of the input signals is provided resulting in final centre channel and at least one LFE sub-woofer channel wherein the problems and disadvantages of the prior art are resolved. The result is a centre and LFE signal that contains a stable, non time-smeared image with a high quality natural-sounding fidelity. These advantages are achieved especially for time-delayed or phase-panned stereo input signals, independently of whether they are matrix encoded or non-matrix encoded input signals.

Abstract: When a material sharing mode using a projector at a first point and the projector at a second point is active, the projector at the first point causes a projection unit to project a video image from a first personal computer onto a projection screen, causes an image taking unit to take an image of the projected video image on the projection screen, causes a control unit of the projector or the first personal computer to compare the taken video image with a supply video image supplied from the first personal computer to the projector, and supplies a difference video image generated based on the comparison to the projector at the second point so as to be superimposed on the supply video image.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: An audio signal having at least one audio channel and associated direction parameters indicating a direction of origin of a portion of the audio channel with respect to a recording position is reconstructed to derive a reconstructed audio signal. A desired direction of origin with respect to the recording position is selected. The portion of the audio channel is modified for deriving a reconstructed portion of the reconstructed audio signal, wherein the modifying includes increasing an intensity of the portion of the audio channel having direction parameters indicating a direction of origin close to the desired direction of origin with respect to another portion of the audio channel having direction parameters indicating a direction of origin further away from the desired direction of origin.

Abstract: A method processing a signal, an encoding apparatus, and a decoding apparatus are provided. The method of processing a signal includes restoring a down-mixed original signal using a re-quantized prediction parameter to generate a restored signal in an encoding apparatus; generating mute information indicating whether the down-mixed original signal has been muted, according to a value of the restored signal; and transmitting the mute information and the down-mixed original signal from the encoding apparatus to a decoding apparatus.

Abstract: Six input channels, to which are allocated 5.1-channel surround signals from a plurality of input sources, are grouped into a surround channel group, and these six input channels are connected to corresponding ones of six surround buses in a one-to-one relationship. Thus, merely grouping the input channels into a surround channel group allows the signals of the individual input channels (5.1-channel surround signals) to be taken out via a plurality of output destinations (5.1-channel speakers) corresponding to the surround buses. Once an instruction is given for changing a value of a parameter, the parameter is controlled in a ganged fashion in all of the input channels of the surround channel group except for an LFE input channel of the surround channel group.

Abstract: A process and system for enhancing and customizing sound includes receiving an input audio sound and enhancing the voice audio input in two or more harmonic and dynamic ranges by re-synthesizing the audio into a full range PCM wave. A tone adjusting circuit is provided which includes a first section for adjusting a low frequency tone, a second section for adjusting a mid frequency tone, a third section for adjusting a high frequency tone and mixing the audio outputs processed by the first, second and third sections to produce an output audio sound. The enhancement includes the parallel processing the input audio via a low pass filter with dynamic offset, an envelope controlled bandpass filter, a high pass filter, adding an amount of dynamic synthesized sub bass to the audio and combining the four treated audio signals in a summing mixer with the original audio. The low frequency tone has a frequency of 100 Hz and a bandwidth of 0.5.

Abstract: An input multi-channel representation is converted into a different output multi-channel representation of a spatial audio signal, in that an intermediate representation of the spatial audio signal is derived, the intermediate representation having direction parameters indicating a direction of origin of a portion of the spatial audio signal; and in that the output multi-channel representation of the spatial audio signal is generated using the intermediate representation of the spatial audio signal.

Abstract: A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.

Abstract: For an audio file that includes multiple channels of audio data, a novel device for detecting the configuration of the audio channels in the multi-channel audio file is presented. The device performs one or more algorithms to determine whether two or more channels are related. Such algorithms are used to distinguish stereo recordings from dual mono recordings. The algorithms are also used to detect any number of related channels, such as distinguishing six related channels from a set of surround sound microphones versus six unrelated channels (e.g., mono or a mixture of stereo and mono audio channels, etc.) These algorithms compare audio channels in pairs in order to determine which channels are sufficiently related as to constitute a stereo pair or a group.

Abstract: A surround sound system comprises a receiver (301) for receiving a multichannel spatial signal that comprises at least one surround channel. A directional ultrasound transducer (305) is used for emitting ultrasound towards a surface to reach a listening position (111) via a reflection of the surface. The ultrasound signal may specifically reach the listening position from the side, above or behind of a nominal listener. A first drive unit (303) generates a drive signal for the directional ultrasound transducer (301) from the surround channel. The use of an ultrasound transducer for providing the surround sound signal provides an improved spatial experience while allowing the speaker to be located e.g. to the front of the user. In particular, an ultrasound beam is much narrower and well defined than conventional audio beams and can accordingly better be directed to provide the desired reflections. In some scenarios, the ultrasound transducer (305) may be supplemented by an audio range loudspeaker (309).

Abstract: A method for encoding a multichannel audio input signal, including steps of generating a downmix of low frequency components of a subset of channels of the input signal, waveform coding each channel of the downmix, thereby generating waveform coded, downmixed data, performing parametric encoding on at least some higher frequency components of each channel of the input signal, thereby generating parametrically coded data, and generating an encoded audio signal (e.g., an E-AC-3 encoded signal) indicative of the waveform coded, downmixed data and the parametrically coded data. Other aspects are methods for decoding such an encoded signal, and systems configured to perform any embodiment of the inventive method.

Abstract: A method and apparatus which encode multi-channel audio signals and a method and apparatus which decode multi-channel audio signals. When encoding, a downmixed audio signal, first additional information for restoring multi-channel audio signals from the downmixed audio signal, and second additional information representing characteristics of a residual signal are multiplexed. When decoding, restored multi-channel audio signals having a predetermined phase difference are combined using the second additional information, and an audio signal of each channel is corrected, in order to improve quality of the restored audio signals.

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

Abstract: One example embodiment increases perceived signal strength of a sound signal based on persistence of hearing. When the ear perceives a signal it takes a finite length of time to process it. During that time period the ear does not recognize input and is, in effect, “Turned Off”. In accordance with one aspect of the technology, these periods of “unheard” audio input are utilized so that they supplement the information which the ear sends to the brain. An exemplary system includes a switch, configured to route the audio alternatively to two signal paths—one which includes a delay circuit that delays that audio by an amount equal to the ear's persistence of hearing interval. The system also includes a signal combiner configured to combine outputs from the two signal paths, so as to provide the brain with a signal that it will perceive as being twice as loud as the original.

Abstract: A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.

Abstract: A surround sound virtualization apparatus and method. The surround sound virtualization apparatus may include an audio decoder to perform head-related transfer function (HRTF) filtering, and a time delay unit to provide a time delay to a plurality of output signals of the audio decoder.

Abstract: This invention relates to reformatting a plurality of audio input signals from a first format to a second format by applying them to a dynamically-varying transformatting matrix. In particular, this invention obtains information attributable to the direction and intensity of one or more directional signal components, calculates the transformatting matrix based on the first and second rules, and applies the audio input signals to the transformatting matrix to produce output signals.

Abstract: An encoder and encoding method for use in a surround sound system wherein at least four audio input signals representing an original sound field are encoded into two channel signals and the encoded two channel signals are decoded into at least four audio output signals corresponding to the four audio input signals. The encoder includes matrix structure connected to receive the four audio input signals for encoding the four input signals into two channel output signals. The matrix structure is responsive to the four input signals for producing L and R output signals as follows: L=FL+kFR+jRL+jkRR R=FR+kFL?jRR?jkRL wherein k denotes a transformation or matrix constant having a value approximately 0.207 and j denotes a 90 degree phase shift.

Abstract: A front surround sound reproduction system which improves the performance of beam steering by using a speaker array arranged geometrically on two or more planes or on one curved surface, and a signal reproducing method of the system. The audio reproduction apparatus to reproduce a multi-channel audio signal by using a plurality of speakers includes a signal distribution unit to duplicate a multi-channel audio signal and to distribute the duplicated signals as one or more groups of multi-channel signals corresponding to one or more speaker array groups, a steering processing unit to form sound beams with steering angles predetermined in relation to each speaker array group, from the groups of multi-channel signals distributed by the signal distribution unit, and a speaker array unit having one or more speaker array groups to reproduce the sound beams of each group formed by the steering processing unit, in the speaker array group.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving a downmix signal including at least one object signal and object information extracted when the downmix signal is generated, receiving mix information including mode selection information, the mix information for controlling the object signal, bypassing the downmix signal or extracting a background object and at least one independent object from the downmix signal, based on the mode selection information, and if the downmix signal is bypassed, generating multi-channel information using the object information and the mix information, wherein the downmix signal corresponds to a mono signal and wherein the mode selection information includes information indicating which one of modes including a normal mode, a mode for controlling the background object, and a mode for controlling the at least one independent object.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving a downmix signal including at least one object signal, and object information extracted when the downmix signal is generated, receiving mix information including mode selection information, the mix information for controlling the object signal, and generating a stereo output signal using the downmix signal or extracting a background object and at least one independent object from the downmix signal, based on the mode selection information, wherein the downmix signal corresponds to a mono signal, wherein the stereo output signal corresponds to a time-domain signal including a signal generated from decorrelating the downmix signal, and wherein the mode selection information includes information indicating which one of modes including a normal mode, a mode for controlling the background object, and a mode for controlling the at least one independent object.

Abstract: A motion picture soundtrack reproduction system has a center front soundtrack channel and a plurality of other soundtrack channels. A volume control adjusts the gain of all the channels. The volume control has a range of settings from a minimum to a maximum, the gain of the center front channel having substantially a first relationship to the volume control settings and the gain of the other channels having substantially a second relationship to the volume control settings, the relationships being such that for a range of volume control settings less than a first setting the gain of the center front channel remains substantially constant while the gain of the other channels decreases as the setting decreases or decreases more gradually than the gain of the other channels as the setting decreases.