Abstract: Systems and methods for providing improved localization of recorded and played back sound are provided by improved microphone arrays for recording sound and by improved systems for playback of sound. Microphone arrays include four microphones with sound transducers located and aimed to mimic capture of sound by human ears. Sound captured by two side-viewing microphones is attenuated, at the time of sound capture and/or recording, at a later processing stage, or at the time of sound playback, by low-pass filtering. The recording maintains four separate channels of sound. Playback occurs through four speakers arranged to reproduce sound in the way human ears hear sound, with appropriate attenuation for side speakers. Playback can also occur through four-channel headphones. Improved playback of two-channel stereo sound can also occur through low-pass filtering of each track and playing the filtered sound through side/rear speakers on the opposite sides.

Abstract: The disclosure relates to a method for processing a stereo signal. The method can include obtaining a center channel signal by up-mixing the stereo signal. The method can also include generating a filtered center channel signal by applying one or more peak filters and one or more notch filters to the center channel signal. Furthermore, the method can include generating a binaural signal based on the filtered center channel signal.

Abstract: Method for generating and outputting an acoustic multichannel signal, comprising the steps of: supplying a stereo signal (S), splitting the supplied stereo signal (S) into a plurality of perception-direction-dependent acoustic signal components (S.1-S.5), generating an acoustic multichannel signal by mixing each perception-direction-dependent acoustic signal component (S.1-S.5) onto an output channel (4.1-4.12) of an acoustic output apparatus (4) that comprises a plurality of, in particular more than two, acoustic output channels (4.1-4.12), outputting the generated multichannel signal over respective acoustic output channels (4.1-4.12) of the acoustic output apparatus (4).

Abstract: A display apparatus including a display panel configured to display an image in a front direction; a main speaker provided on a rear surface of the display panel; and an auxiliary speaker provided on the rear surface of the display panel, the auxiliary speaker being configured to output a sound in a rear direction opposing the front direction.

Abstract: A display apparatus according to an embodiment of the present disclosure includes: a thin plate-like display cell that displays an image; M exciters that are disposed on a back surface side of the display cell, and vibrate the display cell; and a driving section that drives the display cell and the M exciters.

Abstract: An information processing device includes a control unit configured to: acquire information on a destination of a user who uses a resting unit including equipment on which the user is able to sleep, information on an arrival time at the destination, and information on a place in which the user uses the resting unit; transmit a command to a traveling unit configured to be coupled to the resting unit and carry the resting unit such that the traveling unit is coupled to the resting unit and arrives at the destination at the arrival time; and transmit a command to a stimulation device configured to stimulate the user who is using the resting unit such that the stimulation device is activated before the traveling unit is coupled to the resting unit or before the traveling unit departs for the destination after being coupled to the resting unit.

Abstract: A multi-channel and multi-zone audio environment is provided. Various inventions are disclosed that allow playback devices on one or more networks to provide an effective multi-channel and a multi-zone audio environment using timing information. According to one example, timing information is used to coordinate playback devices connected over a low-latency network to provide audio along with a video display. In another example, timing information is used to coordinate playback devices connected over a mesh network to provide audio in one or more zones or zone groups.

Abstract: A phone stand includes a phone holder for coupling to a phone for conducting an audio session, the audio session including at least one voice session conducted by an application executing on the phone and a plurality of microphones including a particular microphone closer to a location where a user is expected to be positioned than other microphones. The phone stand further includes a system controller configured to: receive sound signals from the particular microphone, the sound signals comprising the user's speech; separate the sounds signals into speech signals and non-speech signals; obtain one or more input mixing attributes for the speech signals and the non-speech signals; modify the speech signals and the non-speech signals based on the one or more input mixing attributes; generate mixed signals by combining the modified speech signals and the modified non-speech signals; and send the mixed signals to the phone.

Abstract: A method and system dynamically adjusts the audio of an audio and video signal to improve its overall sound quality and dialog intelligibility. Some embodiments use gain, equalization, audio signal compression and spatial enhancement (reverb) on individual channels of a multichannel audio signal.

Abstract: An audio signal is compressed in an audio coordinate system using gain factors applied in another audio coordinate system. A first component and a second component in a first audio coordinate system is generated from a third component and a fourth component of the audio signal in a second audio coordinate system. An amplitude threshold defining a level for each of the third component and the fourth component for applying compression is determined. A gain factor for the first component is generated using a compression ratio. The gain factor is applied to the first component when one of the third component or the fourth component exceeds the amplitude threshold to generate an adjusted first component. A first output channel and a second output channel in the second audio coordinate system is generated using the adjusted first component and the second component in the first audio coordinate system.

Abstract: Method for generating and outputting an acoustic multichannel signal, comprising the steps of: supplying a stereo signal (S), splitting the supplied stereo signal (S) into a plurality of perception-direction-dependent acoustic signal components (S.1-S.5), generating an acoustic multichannel signal by mixing each perception-direction-dependent acoustic signal component (S.1-S.5) onto an output channel (4.1-4.12) of an acoustic output apparatus (4) that comprises a plurality of, in particular more than two, acoustic output channels (4.1-4.12), outputting the generated multichannel signal over respective acoustic output channels (4.1-4.12) of the acoustic output apparatus (4).

Abstract: A reproduction device according to an embodiment is a reproduction device configured to output a reproduction signal to a speaker system including a first speaker unit group and a second speaker unit group disposed behind the first speaker unit group, the reproduction device including: first delay circuits configured to delay reproduction signals output to speaker units included in the first speaker unit group; and second delay circuits configured to delay reproduction signals output to speaker units included in the second speaker unit group. In the second speaker unit group, a delay time in accordance with a distance difference between a distance and a distance and a longitudinal distance is set for the second delay circuit.

Abstract: A directivity adjustment device that maintains a constant direct-to-reverberant ratio based on the detected location of a listener in relation to the speaker array is described. The directivity adjustment device may include a distance estimator, a directivity compensator, and an array processor. The distance estimator detects the distance between the speaker array and the listener. Based on this detected distance, the directivity compensator calculates a directivity index form a beam produced by the speaker array that maintains a predefined direct-to-reverberant sound energy ratio. The array processor receives the calculated directivity index and processes each channel of a piece of sound program content to produce a set of audio signals that drive one or more of the transducers in the speaker array to generate a beam pattern with the calculated directivity index.

Abstract: The present invention presents an audio triangular system based in the structure of panning in the stereophonic mix, said audio triangular system uses 2 input audio channels and 3 output mono audio playback systems, also creates an assignment of panning values to obtain the audio signals corresponding to each one of the three mono audio playback systems, it is important to mention that the interaction of the audio signals obtained from the previous assignment of values causes the triangular effect of the present invention, additionally, the previously mentioned 3 mono audio playback systems must be capable of reproduce a full frequency range of treble, mid and bass tones each one, furthermore, said mono audio playback systems should be located at a certain distance and angles from each other and in relation to the listener.

Abstract: An audio signal is limited in an audio coordinate system using gain factors applied in another audio coordinate system. A first component and a second component in a first audio coordinate system is generated from a third component and a fourth component of the audio signal in a second audio coordinate system. An amplitude threshold defining a maximum level for each of the third component and the fourth component is determined. One or more gain factors are applied to each of the third component and the fourth component to generate an adjusted third component and an adjusted fourth component, and a first output channel and a second output channel in the second audio coordinate system are generated from the adjusted third and fourth components. The first and second output channels are each limited below the amplitude threshold from application of the one or more gain factors.

Abstract: Disclosed is a loudspeaker array with a multiplicity of array loudspeakers. The multiplicity of array loudspeakers is mounted in a surface of a rigid hemisphere that encloses an acoustic volume. The array loudspeakers are regularly or quasi-regularly distributed over the surface of the rigid hemisphere, and a loudspeaker arrangement includes such an array.

Abstract: In one or more embodiments, one or more systems, methods, and/or processes may receive, via multiple microphones, a first audio signal from a first sound producing source; may convert, via at least one analog to digital converter, the first audio signal to first multiple audio data; may determine a first angle, with respect to a reference, of the first sound producing source based at least on the first multiple of audio data; may determine a sound pressure level based at least on a distance to the first sound source and a type of the multiple speakers; may produce second multiple audio data based at least on the sound pressure level and the first angle; may convert, via at least one digital to analog converter, the second multiple audio data to multiple analog audio signals; and may provide, via the multiple speakers, output audio in a direction of the first angle.

Abstract: An apparatus for generating one or more audio output channel signals depending on two or more audio input channel signals is provided. Each of the two or more audio input channel signals comprises direct signal portions and ambient signal portions. The apparatus comprises a filter determination unit for determining a filter by estimating first power spectral density information and by estimating second power spectral density information. Moreover, the apparatus comprises a signal processor for generating the one or more audio output channel signals by applying the filter on the two or more audio input channel signals. The first power spectral density information indicates power spectral density information on the two or more audio input channel signals, and the second power spectral density information indicates power spectral density information on the ambient signal portions of the two or more audio input channel signals.

Abstract: A method and system dynamically adjusts the audio of an audio and video signal to improve its overall sound quality and dialog intelligibility. Some embodiments use gain, equalization, audio signal compression and spatial enhancement (reverb) on individual channels of a multichannel audio signal.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: A signal processing apparatus for enhancing a voice component within a multi-channel audio signal comprising a left channel audio signal, a center channel audio signal, and a right channel audio signal, the signal processing apparatus comprising a filter and a combiner; wherein the filter is configured to determine an overall magnitude of the multi-channel audio signal over frequency based on the multi-channel audio signal, to obtain a gain function based on a ratio between a magnitude of the center channel audio signal and the overall magnitude of the multi-channel audio signal, and to weight the left channel audio signal, the center channel audio signal, and the right channel audio signal by the gain function; and wherein the combiner is configured to combine individually the left channel audio signal, the center channel audio signal, and the right channel audio signal with the weighted right channel audio signal.

Abstract: A hearing device comprises a processor configured to generate a virtual auditory display comprising a sound field, a plurality of disparate sound field zones, and a plurality of quite zones that provide acoustic contrast between the sound field zones. The sound field zones and the quiet zones remain positionally stationary within the sound field. One or more sensors are configured to sense a plurality of inputs from the wearer. The processor is configured to facilitate movement of the wearer within the sound field in response to a navigation input received from the one or more sensors. The processor is also configured to select one of the sound field zones for playback via a speaker or actuation of a hearing device function in response to a selection input received from the one or more sensors.

Abstract: In one embodiment, a multimedia audio inserter removes the left and right audio channel data from a multimedia signal, for example an HDMI® signal and replaces it with an alternate audio signal, which may be the center channel audio signal provided by a digital surround sound receiver. In exemplary applications, the resulting, modified multimedia signal may be provided to a digital television that is part of a surround sound system, enabling the television to play the center channel signal through its speakers. Optionally, the multimedia audio inserter may provide a copy of the multimedia signal at a multimedia feedthrough output. In exemplary applications, the multimedia feedthrough output may be used to play left and right channel audio, present in the HDMI® signal, through dedicated left and right speakers.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: An apparatus for generating a modified audio signal having two or more modified audio channels from an audio input signal comprising two or more audio input channels is provided. The apparatus has an information generator for generating signal-to-downmix information. The information generator is adapted to generate signal information by combining a spectral value of each of the two or more audio input channels in a first way. The information generator is adapted to generate downmix information by combining the spectral value of each of the two or more audio input channels in a second way being different from the first way. Furthermore, the information generator is adapted to combine the signal information and the downmix information to obtain signal-to-downmix information. The apparatus has a signal attenuator for attenuating the two or more audio input channels depending on the signal-to-downmix information to obtain the two or more modified audio channels.

Abstract: A communicating apparatus comprising a receiving module which comprises: a first receiving line, for receiving a second input data stream; a timing detecting apparatus, for detecting timings for a plurality of data units of the second input data stream; a data allocating apparatus, for allocating the second input data stream to X allocated data streams, wherein the X is a positive integer and the X is larger or equals to 2; and X second receiving lines, for respectively transmitting one of the X allocated data streams.

Abstract: Apparatuses and methods for encoding/decoding scalable digital audio are disclosed. An apparatus for decoding scalable digital audio according to the present invention includes: an audio packet reception unit configured to receive a digital audio packet, including a plurality of pieces of direct audio channel data mapped to respective physical channels and a plurality of pieces of indirect audio channel data indirectly mapped to respective physical channels; an indirect audio channel extraction unit configured to extract the digital audio packet from the pieces of indirect audio channel data; an indirect audio channel decoding unit configured to generate pieces of audio channel data corresponding to a number of physical channels more than the number of pieces of indirect audio channel data using the pieces of indirect audio channel data; and an audio channel output unit configured to match the pieces of audio channel data to the respective physical channels and perform output.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener (4) comprises apparatus (10) to deliver the audio soundfields to the listener (4). The delivery apparatus (10) delivers the audio soundfields to the listener (4) with first and second audio elements perceived by the listener (4) as emanating from first and second virtual source locations (1), respectively, and with the first audio element and/or the second audio element delivered to the listener (4) from a third virtual source location (3). The first virtual source location (1) and the second virtual source location (2) are perceived by the listener (4) as being located to the front of the listener (4), and the third virtual source location (3) is located to the rear or the side of the listener (4).

Abstract: The present invention relates to the entertainment industry and allows to improve the sound quality in audiences for different purposes. The implementation of the present control method is based on the creation of sound clusters, each of them can include any number of nearby located seats equipped with sound reproduction systems. The sound movement through the audience is created by the activation and following deactivation of the acoustic equipment in the mentioned clusters according to the indicated trajectory.

Abstract: A method for processing an audio signal at an audio decoder, the method including receiving a downmix signal, a residual signal, and object information; extracting a background-object signal and a foreground-object signal from the downmix signal using the residual signal and object information; receiving mix information including gain information for the background-object signal; generating a downmix processing information based on the object information and the mix information; and generating an output signal including a modified background-object signal and a modified foreground-object signal. The modified background-object signal is obtained by modifying a gain of the background-object signal using the mix information. The modified foreground-object signal is obtained by modifying a gain of the foreground-object signal using the downmix processing information.

Abstract: An apparatus and method for generating a panoramic sound are provided. The panoramic sound generation apparatus may include a panning coefficient calculation unit to calculate a panning coefficient that represents directivity of a sound source using an input signal, a masker determination unit to determine a direction masker that extracts a sound source of a desired direction based on the panning coefficient, and a channel separation unit to separate to be used as an output signal, output to a sound output device, using the direction masker.

Abstract: A multi-element single enclosure loudspeaker system uses all of the available driver elements in a linear array of loudspeaker drivers for purposes of reproducing center channel program material, whether discrete within a multichannel mix (such as Dolby Digital 5.1™) or derived from a 2-channel mixdown via any appropriate means (such as SRS™ or Dolby ProLogic™ algorithms), in a manner that provides optimized intelligibility of dialog, improved overall clarity, natural timbre and dynamics of music or other effects and wide bandwidth for a wide range of seating/viewing locations for (domestic) home theater environments.

Abstract: An apparatus for providing a current loudspeaker-enclosure-microphone system description of a loudspeaker-enclosure-microphone system is provided. The apparatus has a first transformation unit for generating a plurality of wave-domain loudspeaker audio signals. Moreover, the apparatus has a second transformation unit for generating a plurality of wave-domain microphone audio signals. Furthermore, the apparatus has a system description generator for generating the current loudspeaker-enclosure-microphone system description based on the plurality of wave-domain loudspeaker audio signals, based on the plurality of wave-domain microphone audio signals, and based on a plurality of coupling values, wherein the system description generator is configured to determine each coupling value assigned to a wave-domain pair of a plurality of wave-domain pairs by determining a relation indicator indicating a relation between a loudspeaker-signal-transformation value and a microphone-signal-transformation value.

Abstract: In a method and apparatus for maintaining operation of ancillary equipment associated with a superconducting magnet carrying a DC current, the DC current is directed through a DC-to-AC converter, and the magnitude of the current flowing through the superconducting magnet is ramped down at a controlled rate, thereby generating a controlled voltage across a controlled impedance, and powering the ancillary equipment by the controlled voltage and an associated current, and the ramping rate is controlled in order to maintain a required controlled voltage.

Abstract: An audio processing method and apparatus are described. In one embodiment, at least one first sub-band of a first audio signal is suppressed to obtain a reduced first audio signal with reserved sub-bands; suppressing at least one second sub-band of the at least one second audio signal to obtain at least one reduced second audio signal with reserved sub-bands; and mixing the reduced first audio signal and at least one reduced second audio signal. Alternatively, a first spatial auditory property is assigned to a first audio signal so that the first audio signal may be perceived as originating from a first position. Alternatively, rhythmic similarity between at least two audio signals is detected, and time scaling is applied to an audio signal in response to relatively high rhythmic similarity between the audio signal and the other audio signal(s); and then at least two audio signals are mixed.

Abstract: A method of generating an audio output signal according to a downward compatible sound format, the method including: generating a sum signal by combining a first input channel signal with a second input channel signal; and dynamically correcting the sum signal using samples of the first and second input channel signals from overlapping time windows.

Abstract: A method for pre-processing a channelized music signal to improve perception and appreciation for a hearing prosthesis recipient. In one example, the channelized music signal is a stereo input signal. A device, such as a handheld device, hearing prosthesis, or audio cable, for example, applies a mask to a stereo input signal to extract a center-mixed component from the stereo signal and outputs an output signal comprised of a weighted combination of the extracted center-mixed component and a residual signal comprising a non-extracted part of the stereo input signal. The center-mixed component may contain components, such as leading vocals and/or drums, preferred by hearing prosthesis recipients relative to other components, such as backing vocals or other instruments.

Abstract: A circuit for operating loudspeakers includes a first, second, third and fourth loudspeaker circuit, having one input each for injecting a signal and one output each for connecting a loudspeaker input. The loudspeaker circuits are designed to amplify the injected signal and to provide the amplified signal at the outputs thereof. The loudspeaker circuits can, for example, be used for a 2.1 sound system. The three channels for a 2.1 sound system can be implemented by an amplifier circuit with four loudspeaker circuits, one loudspeaker circuit each being required for the two stereo channels left and right. A subwoofer channel can be driven differentially by two loudspeaker circuits. The stereo channels are, by contrast, only still connected to one loudspeaker circuit each, and so the stereo channels require at least one further common ground cable.

Abstract: A sound system processor for converting left and right channel signals from an audio source into composite left and right signals and employing a mid-side processor, at least one low and high pass filters, and one or more sum processors, together which create at least two or more composite signals for delivery to speaker drivers to generate a broad sound field from a compact multi-speaker sound system source.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal.

Abstract: Provided is a signal processing method which calculates a correlation coefficient indicating the degree of relation in a stereo signal and extracts a speech signal from the stereo signal by using the correlation coefficient and the stereo signal.

Abstract: A driver arrangement for a speaker array (201) comprises a first and second driver (301, 303) each arranged with an on-axis direction at an angle to an on-axis direction of the speaker array (201). The first driver angle exceeds 5° and the second driver angle exceeds the first angle. A front section of each driver (301, 303) comprises a front edge of a radiating element and parts of the driver in front thereof. The drivers (301, 303) are arranged at least partly inline such that a first distance from a front axis (311) perpendicular to an on-axis of the speaker array (201), and intersecting a furthest forwards part of the first driver front section, to a closest part of the second driver front section is lower than a second distance from the front axis (311) to a furthest part of the first driver front section. The combination of angled drivers and the inline arrangement provides improved performance.

Abstract: The invention generally relates to acoustic transducers. In certain aspects, the acoustic transducer includes a diaphragm and an actuator. In further aspects, the acoustic transducer includes a connector having a planar proximal end that tapers to a distal end. The proximal end may be coupled to the diaphragm and the distal end coupled to the actuator, such that the actuator causes movement of the diaphragm. In additional aspects, the acoustic transducer includes a connector having a proximal end that couples to the diaphragm and a distal end that couples to the actuator. The proximal end of the connector can include a damping element. In other aspects, the diaphragm includes a connector, in which the connector tapers to a distal end that is configured to mate with the actuator such that the actuator causes movement of the diaphragm.

Abstract: The invention generally relates to acoustic transducers. In certain aspects, the acoustic transducer includes a speaker in the base of the transducer for emitting acoustic energy at a first range of frequencies. The transducer may also include a diaphragm coupled to the base that is configured to emit acoustic energy at a second range of frequencies. Transducers of the invention may also include at least one actuator coupled to the diaphragm to cause movement of the diaphragm and thus, the acoustic energy at the second range of frequencies.

Abstract: A sound separation device includes: a signal obtainment unit which obtains a plurality of acoustic signals including a first acoustic signal and a second acoustic signal; a differential signal generation unit which generates a differential signal that is a signal representing a difference in a time domain between the first acoustic signal and the second acoustic signal; an acoustic signal generation unit which generates, using at least one acoustic signal among the acoustic signals, a third acoustic signal; and an extraction unit which generates a frequency signal by subtracting, from a signal obtained by transforming the third acoustic signal into a frequency domain, a signal obtained by transforming the differential signal into a frequency domain, and generates a separated acoustic signal by transforming the generated frequency signal into a time domain.

Abstract: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: An audio system receives a multi-channel signal which is fed to a controller (121) that generates a first drive signal for a first sound emitter (111) by combining signals of a plurality of the channels. The first drive signal has a signal component contribution from a first bandwidth of each channel of the multi-channel signal. The multi-channel signal is also fed to another controller (115) which generates second drive signals for second sound emitters (101-109). The second drive signals are generated from a single channel signals of the multi-channel signal and in a second bandwidth having a lower cut-off frequency which is above 950 Hz for a 3 dB gain attenuation relative to an average gain for a frequency band extending 1 kHz above the lower cut-off frequency and higher than a lower cut-off frequency of the first bandwidth. A delay processor (125) introduces a delay for signal components of the first drive signal relative a corresponding second drive signal.

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.

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.