Abstract: A method of down mixing a plurality of signals including at least a left surround signal, a right surround signal, a left signal and a right signal into a stereophonic pair that is a left transmitted signal and a right transmitted signal includes summing the left surround and right surround signals to provide a monophonic surround signal, filtering the monophonic surround signal to provide a filtered monophonic signal having properties related to the diffraction pattern around the head of a listener, multiplying the filtered monophonic surround signal with a right coefficient signal related to the ratio of the magnitude of the right surround signal to the sum of the magnitudes of the left surround signal and the right surround signal to provide a left product signal and multiplying the filtered monophonic surround signal with a left coefficient signal related to the ratio of the magnitude of the left surround signal to the sum of the magnitudes of the left surround signal and the right surround signal to prov

Abstract: Surround sound encoding and decoding includes apparatus and techniques for encoding a number of major channels of a surround sound signal, such as left, right, center, left surround and right surround, into a lesser number of channels, such as left transmitted and right transmitted, and decoding the lesser number of channels into a larger number of decoded channels, such as left, right, center, left surround and right surround, for being electroacoustically transduced.

Abstract: A apparatus formed on a motherboard of computer complied with PC99 standard for converting to six-channel output from two-channel is disclosed. The apparatus comprises a CODEC, a MIC_IN connector switch, first, second, and third filters, and a resistor circuit. In a six-channel output mode, the rear left and rear right surround signals and LINE_IN_L and LINE_IN_R signals are fed into the third filter.

Abstract: A method for realizing virtual multi-channel by spectrum analysis is disclosed. The low-frequency-effect sound data is obtained by averaging sound data of left and right channels in the two-channel sound source and filtering the average by a first low pass filter. The central channel sound data is obtained by averaging sound data of the left and the right channels in the two-channel sound source and filtering the average by a second low pass filter. The surround right and left sound data is obtained by copying sound data of the left and the right channels in the two-channel sound source, respectively.

Abstract: Reproduction accuracy of a digital signal, representing for example stereo audio signals, is improved by transmitting sample data which have been encoded to form transmission signals. The transmission signals are arranged in consecutive frames, each frame including a plurality of information packets, and each information packet including N bits. The number of information packets in a frame is determined by the ratio of the product of the transmission bit-rate and the number of samples represented in the frame, to the product of the number of bits in a packet and the sampling frequency. The transmission signals are recorded in a record carrier, or are transmitted in real time. In a digital stereo audio signal embodiment, the samples may be sub-band encoded or transform encoded. The transmission signal may include an indicator signal indicating a combination of certain samples, or scale factor signals for specific components of the sampled signal.

Abstract: The beginning detection, accommodation and frequency bias properties of the human hearing mechanism have been modeled to create systems that can detect directional transients (“sound events”) in a sound field and separately localize them. These models break down a sound field into sound events and non-sound events and separately localize the sound events and non-sound events. Sound events are generally identified according to the frequency bias and beginning detection properties. Once detected, sound events are generally localized according to differential steering angles (steering angles to which the steady-state signals have been accommodated) or ordinary steering angles and which reflect the direction of a sound event indicated in its rise-time. When no sound events are detected, non-sound events are localized according to a steering angle that does not reflect rapid motion.

Abstract: A method and apparatus for preserving matrix-surround information in encoded audio/video includes a receiver operative to receive matrix-surround encoded audio signals via a modem, separate the audio signals into a frequency spectrum having discrete audio frequencies, and determine a cutoff threshold used to encode the matrix-surround encoded audio signals. The method and apparatus further includes a decoder operative to decode a first set of the audio frequencies below the determined cutoff threshold using a first matrix-surround preserving audio encoding method and to decode a second set of audio frequencies above the cutoff threshold using a second non matrix-surround preserving audio encoding method.

Abstract: A technique for panning over four speakers utilizes an internal three-channel representation (Left, Right, W) where all channel weights are positive. The three channels are combined by a decoder to drive pairs of front and back speakers.

Abstract: Methods and apparatus for recording on DVD-like recording media in which audio content is stored in a high-capacity multi-channel (e.g., six-channel) format are provided. Various channels may use various resolutions. A two-channel audio output may be derived from the multi-channel audio data stream during playback. To facilitate an accurate derivation, the mixing coefficients to be used in generating the derivation can be supplied along with the six-channel audio data.

Abstract: The method and system of present invention sequences audio post-processing algorithms to simulate live or theater sound. An audio signal is selectively post-processed according to equipment availability and listener preferences. Downmixing or Prologic algorithms are applied to a signal arriving at sound system. A listener inputs their speaker configuration to a player console. Desired post-processing effects are likewise indicated to the console. For instance, if surround sound equipment is both available and selected, then surround portions of the audio signal are parsed to surround speakers. Bass management techniques then transfer low frequency channels of the signal to compatible speakers. VES or DCS algorithms further manipulate the surround portion of the signal to create an illusion of immersion, and a center channel equalizer balances the signal playback. Alternatively, the post-processed signal is transmitted to a headphone set.

Abstract: The pitch extracting part generates a pitch waveform signal in a manner making the time interval of the pitch of the input audio sound data to be the same. After the number of samples in each region is made to be the same by the re-sampling part, the pitch waveform signal is changed into a subband data that express a time-varying-strength of a basic frequency composition and a higher harmonic composition by the subband analyzing part. The subband data are superimposed by a modulation wave composition that expresses attaching data of an attaching object by the data attaching part and is regarded as a bit stream to output through a nonlinear quantizing. A portion expressing the higher harmonic composition that is made corresponding to the audio sound expressed by this audio sound data in the subband data are deleted by the encoding part.

Abstract: An apparatus and method for assigning each conferee to a conference a three-dimensional position with respect to a central listening position and the other conferees. Each conferees audio stream is encoded with the assigned three-dimensional position to produce an encoded audio stream corresponding to each conferee. For each conferee, the encoded audio streams of the other conferees are mixed to produce a mixed audio stream wherein the conferee listens to the conference from the central listening position.

Abstract: The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi−1 for an encoding unit Ai−1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

Abstract: A scalable stereo audio encoding/decoding method and apparatus are provided.

Abstract: An audio processing unit that provides a high-sound-quality reproduction even when a band property of an input encoded signal drops at or below a Nyquist frequency. The audio processing unit has a band-variable bandpass filter 1500; a band determining section 1700 for determining a passband of the band-variable bandpass filter with respect to a band extension component from an extended band generating section 1300 by using decoding information from a decode-processing section 1100 as a band determination information; and a bandpass filter controller 1600 for controlling the passband of the band-variable bandpass filter in accordance with an indication from the band determining section.

Abstract: A method and system for coding audio signals in a multi-channel sound system, wherein a plurality of MDCT units are used to reduce the audio signals for providing a plurality of MDCT coefficients. The MDCT coefficients are quantized according to the masking threshold calculated from a psychoacoustic model and a plurality of INT (integer-to-integer) DCT modules are used to remove the cross-channel redundancy in the quantized MDCT coefficients. The output from the INT-DCT modules is Huffman coded and written to a bitstream for transmission or storage.

Abstract: An L-channel signal and an R-channel signal are input into a subtractor in which the difference between the L-channel signal and the R-channel signal is determined. The resulting signal is input into first and second multipliers. In the first multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a previous frame. In the second multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a current frame. The mixed output signals from the first and second multipliers respectively undergo spectrum transform in first and second spectrum transformers, and quantized in first and second quantizers. A comparator compares a quantization error in the first quantizer with that in the second quantizer. If the quantization error in the second quantizer is greater than that in the first quantizer by a predetermined factor, the comparator controls a switch to select the output of the first quantizer.

Abstract: A method and system for reconfiguring a multi-dimension sound system is disclosed. The method and system include a remote control that is capable of emitting a low-frequency sound wave pulse pattern, and a plurality of speakers, wherein each speaker includes a pulse counter that counts a number of sound wave pulses received from the remote control over a predetermined time period. The method and system further include an audio receiver that includes a digital audio encoding system. The receiver uses the respective sound wave pulse count from each of the speakers to compute the distance between the remote control and each speaker, and programs the digital audio encoding system with speaker delay times based on the computed distances.

Abstract: A conveyor mat built up from hingedly intercoupled plastic modules, each module consisting of a number of substantially equally spaced apart hinge plates extending throughout the length of the module and provided with hinge loops at both ends thereof, the outer hinge plate and a plate-shaped part located on the outer side of a module of at least a number of modules extending to a position below the bottom face of the mat formed by the successive modules or being provided with a recess in the longitudinal direction of the module, the arrangement being such that the thus formed downward extension and/or the recesses form a lateral guide for the conveyor mat and can cooperate with a tubular or U-shaped guide section respectively.

Abstract: Scalable coding of audio into a core layer in response to a desired noise spectrum established according to psychoacoustic principles supports coding augmentation data into augmentation layers in response to various criteria including offset of such desired noise spectrum. Compatible decoding provides a plurality of decoded resolutions from a single signal. Coding is preferably performed on subband signals generated according to spectral transform, quadrature mirror filtering, or other conventional processing of audio input. A scalable data structure for audio transmission includes core and augmentation layers, the former for carrying a first coding of an audio signal that places post decode noise beneath a desired noise spectrum, the later for carrying offset data regarding the desired noise spectrum and data about coding of the audio signal that places post decode noise beneath the desired noise spectrum shifted by the offset data.

Abstract: The present invention comprises methods and systems for a dynamic audio processor for processing an audio signal prior to encoding. The audio signal is pre-processed to provide analog-to-analog modification that creates a preferred analog format that is effectively and efficiently ready to be converted into a digital data stream by an encoder, such as an A-to-D converter or codec. One possible application includes, for example, a web streamer that provides digital data streams from a server. The inventive arrangements are preferably carried by functional stand-alone components, integral components of a PC or other computing platform, or carried as part of the encoder. Preferably, the pre-processing comprises receiving the audio signal, nominalizing the signal to provide a level input, and compressing and equalizing the signal for outputting thereof.

Abstract: A sound signal encoding apparatus for encoding two different sound signals, comprising: compression level calculating means for calculating a compression level for each of the sound signal sections; compression level judging means for judging whether or not the calculated compression level for each of the sound signal sections exceeds a predetermined threshold compression value; frequency components encoding means for encoding the quantized frequency components for each of the sound signal sections to a multiplexed bit stream with a predetermined bit rate under two different states consisting of the first state in which the frequency components for each of the sound signal sections are compressed by the first frequency components compressing means when the compression level judging means is operative to judge that the compression level for each of the sound signal sections exceeds the predetermined threshold compression value and the second state in which the frequency components for each of the sound signal

Abstract: An audio transmission system is provided which is operable to receive sets of signals representative of a speech signal generated by a speech source. The system is operable to determine parameters representative of each of the sets of signal values, to encode these parameter values and then to transmit these parameter values to a receiver circuit. At the receiver, the encoded parameters are decoded to recover the parameter values. The recovered parameter values may then be used to resynthesise the speech using a speech synthesiser or they may be used by a speech recognition system to generate a recognition result.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: In the coding and decoding of stereo audio spectral values both the intensity stereo process and prediction are used in order to achieve high data compression. If intensity stereo coding is active in one section of scale factor bands, the prediction for the right channel in that range is deactivated, whereby the results of the prediction are not used to form the coded stereo audio spectral values. To allow further adaptation of the prediction for the right channel, the predictor of the right channel is fed with stereo audio spectral values for the channel, which again are intensity stereo decoded.

Abstract: The invention provides a sound field correction circuit for a surround playback apparatus wherein a decoding circuit is formed from a comparatively simple circuit and similar effects to those of a ordinary sound field correction circuit can be achieved at a reduced cost. A sound field correction circuit includes a decoding circuit for restoring 2-channel stereo signals encoded for a multi-channel surround effect to obtain multi-channel surround signals. The decoding circuit includes an adder for adding the encoded 2-channel stereo signals to each other, a subtractor for subtracting one of the encoded 2-chhanel stereo signals from the other, a level adjusting volume for adjusting the output level of the adder, and another level adjusting volume for adjusting the output level of the subtractor. An output of the level adjusting volume is used as a center channel signal while an output of the level adjusting volume 6 is used as a surround channel signal.

Abstract: For audio coding, the MPEG-2 standard provides for a spatial representation by multi-channel reproduction. In order to ensure backwards compatibility with MPEG-1 signals, the signals of the multi-channel sound channels are matrixed. Before being matrixed, the audio signals generally have their levels reduced in order to prevent overdriving. In order to balance again the reduction at the encoder end, an option that is provided in the MPEG-2 standard is for a decoder to raise the output levels again. If, however, a transmission error then occurs in the multi-channel section, the decoder cannot carry out the reverse matrixing. In that case, only the MPEG-1-compatible signal component will be decoded. The auditory impression can therefore be disturbed in various ways during the changeover to MPEG-1 decoding.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A matrix system encodes five discrete audio signals down to a two-channel stereo recording and decodes the recorded stereo signal into at least five stand alone, independent channels to allow placement of specific sounds at any one of 5 or more predetermined locations as individual, independent sound sources, thus producing a 5-2-5 matrix system. One embodiment of the system provides signals to left front, right front, center, left rear, and right rear speaker locations. The matrix system is compatible with all existing stereo materials and material encoded for use with other existing surround systems. Material specifically encoded for this system can be played back through any other existing decoding systems without producing undesirable results.

Abstract: The system and method of the present invention provides enhanced surround sound effects. In one embodiment, 90.degree. phase shift and copy of the modified head related transfer functions (HRTF) are applied to each rear sound signal. The modified HRTF for each rear signal is generated by removing the head related transfer function corresponding to the front center signal from the HRTF corresponding to the rear sound signal. This provides the audible effect of distinguishing more clearly sounds originating in front of the listener or to the rear of the listener while not limiting the perceived bandwidth of the signal. The rear signals corresponding to the first channel are inverted. The front and rear signals for each corresponding channel are then combined. The two channels, generated can be stored on media, such as film, and subsequently read and input to a surround sound decoder to generate enhanced surround sound output.

Abstract: An apparatus for encoding a plurality of n digital information signals is provided with at least a first, second and third input terminal (1,2,3) for receiving a first, a second and a third digital information signal (L,R,C). A matrixing unit (10) is provided for generating a first and a second digital composite signal (L.sub.0,R.sub.0) from the at least the first, second and third digital information signals. First and second data compression means (22.1,22.2) are provided for data compressing the first and second composite signal (L.sub.0,R.sub.0) so as to obtain first and second data reduced digital composite signals. A first selection unit (16) is provided for selecting at least one information signal from said plurality of n information signals so as to obtain at least a first auxiliary signal (AUX.sub.1). At least a third data compression means (22.3) is present for data compressing the at least first auxiliary signal so as to obtain a first data reduced auxiliary signal.

Abstract: A decoder capable of decoding audio signals encoded using analog noise reduction techniques, particularly Dolby SR, is implemented by means of digital signal processing. The typical decoding paradigm used by analog Dolby SR decoders is modified to overcome the problems caused by delay which is introduced by the digital implementation. A gain factor is factored out of the feedback transfer function and is placed in the forward signal path of each stage of the digital decoder. A single control circuit can then be used to generate coefficients which define transfer functions of each filter stage of the signal path. Linear interpolation techniques are used to limit the number of coefficient entries necessary to emulate the transfer function of an analog Dolby SR decoder. The digital decoder structure can be inverted to produce a digital encoder which is capable of implementing any noise reduction standard desired, including Dolby SR.

Abstract: A sound reproduction system for converting stereo signals on two input channels, which may have been directionally encoded from a four or five channel original using a phase/amplitude film matrix encoder, such signals including at least one component which is directionally encoded through a phase and amplitude encoding device and at least one component that is not directionally encoded but is different in the two input channels, into signals for multiple output channels, for example center, front left, front right, side left, side right, rear left, and rear right, including decoding apparatus for enhancing the directionally encoded component of the input signals in the desired direction and reducing the strength of such signals in channels not associated with the encoded direction, while preserving both the maximum separation between the respective left and right channels and the total energy of the non-directionally encoded component of the input channels in each output channel, such that the instruments rec

Abstract: A sound system is disclosed which, in common with earlier phase-amplitude multichannel "matrix" encode-decode systems, conveys or stores audio programs having multidirectional sound-source localization in a pair of audio-bandwidth channels, whether analog or digital. In the present invention, representation of a vertical, height dimension is added by mapping the "phase-amplitude sphere" representing signal separation onto a spatial hemisphere and by introducing to the parameters of phase difference and amplitude ratio a third parameter, decorrelation. Non-complementary matrices are used for encoding and decoding to provide improved separation between decoded signals. A radius-scaling function facilitates encoding of sound source locations outside, as well as within, the boundaries of the audience space defined by the peripheral and overhead loudspeaker locations.

Abstract: A 7-channel encoder and corresponding decoder is disclosed for encoding a 7-channel signal into a transmission signal which is backwards compatible, so that prior art MPEG-1 2-channel decoders are capable of decoding the transmission signal into a compatible stereo signal, prior art MPEG-2 5-channel decoders are capable of decoding the transmission signal into a compatible 5-channel signal. Further, a 7-channel decoder is disclosed for decoding the transmission signal into a 7-channel signal. (FIG. 2).

Abstract: A method for reducing data during the transmission and/or storage of the digital signals of several dependent channels is described in which the dependence of the signals in the channels, e.g. in a left and a right stereo channel, can be used for an additional data reduction. Instead of known methods such as middle/side encoding or the intensity stereo process that lead to perceptible interference in the case of an unfavourable signal composition, the method according to the invention avoids such interference, in that a common encoding of the channels only takes place if there is an adequate spectral similarity of the signals in the two channels. An additional data reduction can be achieved in that in those frequency ranges where the spectral energy of a channel does not exceed a predeterminable fraction of the total spectral energy, the associated spectral values are set at zero.

Abstract: An improved theater surround sound system incorporates a screen top speaker and an overhead speaker driven by corresponding upper surround channels to more accurately reproduce sounds produced from above the listener. These top surround channels are encoded along with the left and right surround channels. A pilot signal is used to direct the sound from the side surround channels to the upper surround channels when needed.

Abstract: A sound reproduction system for converting stereo signals on two input channels, which may have been directionally encoded from a four or five channel original using a phase/amplitude film matrix encoder, such signals including at least one component which is directionally encoded through a phase and amplitude encoding device and at least one component that is not directionally encoded but is different in the two input channels, into signals for multiple output channels, for example center, front left, front right, side left, side right, rear left, and rear right, including decoding apparatus for enhancing the directionally encoded component of the input signals in the desired direction and reducing the strength of such signals in channels not associated with the encoded direction, while preserving both the maximum separation between the respective left and right channels and the total energy of the non-directionally encoded component of the input channels in each output channel, such that the instruments rec

Abstract: A recording and reproducing system is arranged to record converted two-channel audio signals obtained by using all four-channel stereophonic audio signals and audio signals of two specific channels included in the four-channel stereophonic audio signals on a recording medium as four-channel recording audio signals and, at the time of reproduction, to be capable of restoring the four-channel recording audio signals to the four-channel stereophonic audio signals and also to permit either the four-channel stereophonic audio signals or the converted two-channel audio signals to be selectively outputted.

Abstract: A system for controlling distribution of sound in a multi-channel sound system such as a "surround sound" system in a movie theater. Control signals are used to steer, or distribute, portions of the sound signal between designated speakers. In a preferred embodiment the control signals are provided with the sound signals, themselves, and are separated from the sound signals by using a pilot signal or signals.

Abstract: In a method of coding a plurality of audio signals, the left and the right basic channel as well as the central channel are combined by joint stereo coding so as to obtain a jointly coded signal, which is decoded so as to provide simulated decoded signals. The simulated decoded signals and two surround channels are combined by matricization by means of a compatibility matrix so as to form compatible signals which are suitable for decoding by existing decoders.

Abstract: An adaptive high-fidelity perceptual subband encoder for spatially directional multichannel audio signals, using a mixture of individual-channel subband coding with subband-steered composite-channel signal coding that combines the spectral components of the individual channel subbands selected for steering. The composite channel representation includes a subband steering control signal which either conveys the levels of the spectral components from all the steered channels, or represents the apparent directions (net directional vectors) of the spectral components from all the steered channels.

Abstract: An improved theater surround sound system incorporates a screen top speaker and an overhead speaker driven by corresponding upper surround channels to more accurately reproduce sounds produced from above the listener. These top surround channels are encoded along with the left and right surround channels. A pilot signal is used to direct the sound from the side surround channels to the upper surround channels when needed.

Abstract: A method of reproducing a sound field using digital signal processors in which 2-channel stereo signals are received and converted into 4-channel sound field signals through a first DSP and the 4-channel sound field signals are adaptively composed corresponding to a headphone mode or a speaker mode, again to be into 2-channel sound field signals through a second DSP thereby reproducing 4-channel sound field feeling by 2 channels.

Abstract: A transmission channel may be utilized for a combination of wide-band signals and auxiliary signals, while still providing improved reproduction accuracy. Digital sub-signals or subband signals representing at least first and second data-reduced components are combined with at least one auxiliary digital signal in defined frame portions before transmission. The auxiliary signal may be user-discernible information such as third and fourth audio channels, so that a four channel surround sound signal is transmitted; or teletext or video or picture information may be provided. Sampling reduction and filtering into subbands may be applied to the auxiliary signal as well as to the first and second signal components.

Abstract: A method and apparatus for dynamic, adaptive mapping of three-dimensional aural phenomenon to a two-dimensional sound reproducing surface. The surface is comprised of sound pixels that are discrete, addressable output locations. Actual or synthetic recorded audio material is decomposed into discrete sound sources which are then mapped to the sound pixels. This mapping occurs during the production phase by means of computer-aided design (CAD) system. The CAD system guides the sound designer in the implementation of complex acoustical designs by automating the most complex and computationally intensive functions. The mapping function can be extended to integrate real-time human interaction, ultimately creating a participatory, interactive virtual sound environment.

Abstract: In a system in which a low-bit rate encoder and decoder carries matrixed audio signals, an adaptive rematrix rematrixes matrixed signals from an unmodified 4:2 matrix encoder to separate and isolate quiet components from loud ones, thereby avoiding the corruption of quiet signals with the low-bit-rate coding quantization noise of loud signals. The decoder is similarly equipped with a rematrix, which tracks the encoder rematrix and restores the signals to the form required by the unmodified 2:4 matrix decoder. The encoder adaptive rematrix selects the matrix output signals or the amplitude weighted sum and difference of the matrix output signals. The choice of whether the matrix output signals or the sum and difference of the matrix output signals are selected is based on a determination of which results in fewer undesirable artifacts when the output audio signals are recovered in the decoder.

Abstract: Left right and surround components of a stereo signal are coded into a monaural and small co-channel providing volume steering for recreating a stereo effect with a substantially reduced bit rate. The signal is split into a sum and difference signal, the difference signal is randomized and the sum is added to the randomized difference to comprise the single audio channel. A functional relationship is solved for left and right volumes which is then transmitted for intervals on the co-channel. Decoding of the single transmission channel directs it to left, right, or surround channels based on decoding on the logic co-channel. The co-channel updates left and right volume levels which are interpolated through time to effect smooth volume change. Surround gain is determined from left and right channel gains to maintain unity total volume, with the sum of the squares of the three volume controls being unity.

Abstract: A television receiver includes a volume control circuit (4) for controlling levels of sound signals corresponding to a left channel, a right channel, a center channel and a surround channel and speakers (6 to 10) corresponding to these channels. The television receiver further comprises a test tone circuit (11; 18), a microcomputer (14) and a character display circuit (16). The test tone circuit (11; 18) supplies a test tone of a prescribed frequency sequentially to the speakers (6 to 10) through the volume control circuit (4). The microcomputer (14) provides a command to the character display circuit (16) so that a CRT 17) displays which speaker is currently supplied with the test tone as well as volume levels of the respective speakers.

Abstract: A signal encoding system for generating at least two encoded signals in response to a plurality of information signals containing audio data representative of virtual images of sources of the data so that the encoded signals can be subsequently decoded by a decoding system including a decoding matrix means for decoding the encoded signals so as to regenerate the information signals. The signal encoding system comprises signal encoding means for generating the encoded signals in response to the information signals; and a duplicate of the decoding matrix means coupled to the encoding means for generating a feedback signal corresponding to each of the information signals; wherein the encoding signals are modified in response to the feedback signals so as to reduce the noise artifacts generated by the decoding matrix when the encoded signals are decoded by the decoding system.