Abstract: There provided a sound data delay unit for delaying sound data of a plurality of channels in a single memory area, the sound data delay unit wherein, when delay amount assigned to each channel is input into an address information generator, the address information generator generates address information for assigning storage area of the sound data for each channel in a memory, and a selector inputs the address information into the memory with respect to each channel and assigns the storage area for each channel.

Abstract: An external input audio processor includes an input terminal for inputting an audio stream, an input terminal for inputting an audio serial data, an input terminal for inputting digital data obtained by analog-to-digital conversion of an analog input, an input terminal for inputting external digital audio data, a selector for selecting one of the digital data supplied from the four input terminals, and an audio processing circuit for carrying out audio processing of the digital data selected by the selector. The external input audio processor can reduce the circuit scale of the system.

Abstract: A matrix surround decoding system comprises a DIR 10 detecting a sampling frequency FS included in a stereophonic source, and a microcomputer 12 determining the type of stereo source on the basis of the sampling frequency detected by the DIR 10, and outputting to a matrix surround decoder 11 a mode switching signal CL for switching the output mode of the matrix surround decoder 11 to a mode corresponding to the determined type of stereophonic source.

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: In a multi-channel acoustic apparatus with a DVD reproduction function, comprising a DVD reproduction control unit for controlling the reproduction of a DVD, and a system control unit for controlling the overall acoustic apparatus and controlling the reproduction of another input source other than the DVD, the DVD reproduction control unit comprising the function of controlling a first Dolby Pro Logic decoder which can be used at the time of reproducing the DVD, and the system control unit comprising the function of controlling a second Dolby Pro Logic decoder which can be used at the time of reproducing the input source other than the DVD, there are provided means for adjusting a surround delay time period on the basis of a user operation, and means for transmitting to the system control unit the adjusted surround delay time period.

Abstract: An input digital audio signal is divided into sub-band signals in respective sub-bands. Scale factors of the respective sub-bands are determined on the basis of the sub-band signals for every frame. Calculation is made as to differences between the determined scale factors for a first frame and the determined scale factors for a second frame preceding the first frame. Absolute values of the calculated scale-factor differences are calculated, and data representative of the calculated absolute values are generated. The data representative of the calculated absolute values are encoded into data of a Huffman code. Sign bits are generated which represent signs of the calculated scale-factor differences. The sub-band signals are quantized in response to the determined scale factors for every frame to generate quantized samples of the sub-band signals. The Huffman-code data, the generated sign bits, and the quantized samples of the sub-band signals are combined into a bit stream.

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: A system and method are disclosed for performing digital multi-channel decoding of a BTSC composite audio signal. Each subsequent stage of the digital multi-channel decoding process is performed at the lowest sampling rate that yields acceptable performance for that stage. Analog-to-digital conversion of the composite audio signal is performed first to generate a composite digital audio signal. After analog-to-digital conversion, all signal processing may be performed in the digital domain. The composite digital audio signal is digitally filtered to frequency compensate for variations caused by previous stages of processing, including IF demodulation. Digital channel demodulation and filtering are performed to isolate single channels of the composite digital audio signal such as SAP, L−R, and L+R channels.

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

Abstract: The audio decoding apparatus comprises a CPU which groups received plurality of sample data into one block. Furthermore, the CPU adds control information relating to attributes to data of each block.

Abstract: A PC audio mixer configured to mix a PC audio signal having one or more PC audio signal channels with an audio surround sound signal having three or more audio surround sound channels, and generate an output signal having at least three output channels which are functions of said PC audio signal and the audio surround sound signal. Preferably, the PC audio signal is an audio signal which is derived from a PC data signal. The PC data signal may comprise a MIDI, WAVE, AIFF, AU, or an MP3 data signal.

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: A digital audio decoder decodes or expands compressed data such as bit stream data, which are compressed based on the MPEG/Audio standard. Inverse quantization circuits perform inverse quantization on plural bit stream data, which are supplied thereto in connection with multiple channels respectively, thus producing inversely quantized data with respect to a prescribed number (e.g., thirty two) of sub-band samples respectively. The inversely quantized data are combined together among the multiple channels with respect to the prescribed number of the sub-band samples respectively. Then, a filter bank synthesizes together combined data corresponding to all of the sub-band samples, thus reproducing original digital audio signals. Multipliers are provided for use in gain control on the inversely quantized data with respect to the sub-band samples respectively.

Abstract: An apparatus for recording a multichannel program includes an audio converter adapted to convert said multichannel program at least partially into a modified program requiring less channels than said multichannel program. The audio converter including control elements used to modify the conversion process used to generate said modified program. A coefficient generator coupled to said converter to generate a coefficient indicative of the operation of said controls. A mixer is arranged to mix the coefficient with program data. A recorder is arranged to record the mixed data on a medium, whereby the medium can be played either using a multichannel player device or a reduced channel player device by using said coefficient.

Abstract: An architecture allows two-channel sound effect hardware to output four-channel analog signal, comprising sound effect application program for processing multi-channel information of application software. An operation system kernel program is received the multi-channel information by a multi-channel sound effect interface. Means for transforming a four-channel signal to two-channel signal is coupled to the operation system kernel program to transform a four-channel information to a two-channel information. A two-channel sound effect hardware is coupled to the means for transforming a four-channel signal to two-channel signal. A converter is coupled to the two-channel sound effect hardware to transform the two-channel signal to a four-channel analog signal.

Abstract: A signal processing apparatus includes a decoder for decoding a stream signal so as to generate a digital audio signal of a low frequency effect channel and digital audio signals of first through n'th (n≧2) channels; an adder section for adding the digital audio signal of the low frequency effect channel and the digital audio signal of a specified channel among the first through n'th channels, so as to generate an addition signal; an n number of D/A conversion sections for converting the digital audio signals of the first through n'th channels, excluding the digital audio signal of the specified channel, and the addition signal into n types of analog audio signals; a first signal processing section for generating a digital audio signal of the low frequency effect channel; and a second signal processing section for generating an analog audio signal of the specified channel.

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: A digital information reproduction apparatus for compressing frames of data so as to intermittently reproduce the data in fast-forward speed. A data re-write device retrieves a reproduction-target frame for the fast-forward reproduction and a parameter-calculation-target frame exclusively for calculation of the frame data parameters. The frames then are transferred to a RAM. A decoder determines the frame data in the RAM, calculates the parameter-calculation frame data, and decodes the reproduction frame data by use of the parameter obtained as above. The described operation prevents expending the reproduction frame data with an incorrect parameter so as to normally fast-forward reproduce the compressed data. Furthermore, the present can select a frame used in the fast forward reproduction according to a feature index so that a section subjected to the fast-forward reproduction can be easily grasped by the user.

Abstract: In a digital audio system, a first audio unit has its audio decoder adapted to decode coded digital audio data to get coding parameters including information of frame borders of the audio data and release decoded digital audio data and coding parameters. A second audio unit which receives the digital audio data and coding parameters has its audio encoder adapted to re-encode the digital audio data by using the coding parameters. The re-encoding process by use of the best or highly optimized coding parameters, which have been used for original coding, restores the same state of audio data, and the reduction of coding errors can be expected.

Abstract: An improved method and apparatus for recreating spatially natural soundfields in a variety of listening environments is disclosed. The apparatus includes a speaker cabinet that contains multiple drivers each driven so as to effect an overall radiation pattern that mimics the soundfield pattern described by a B-format signal vector. This apparatus allows natural or synthetic soundfields to be played back from a relatively small mechanical speaker structure, providing the benefits of multichannel sound without a cumbersome arrangement of individual speaker cabinets.

Abstract: The invention concerns an apparatus for decoding an enhanced stereo sound effect from a stereo signal. The apparatus has an input for receiving a stereo signal including localisation cues, the stereo signal comprising a left channel and a right channel, each of the left and right channels being carried over a pair of wires including a positive lead and a negative lead. The positive leads of the right and left channels are outputted directly. A decoding circuit for decoding the localisation cues from the stereo signal is placed on the negative leads of the right and left channels. The apparatus of the present invention does not require an independent power source and uses the information contained in the original signal to produce and enhanced signal.

Abstract: The present invention was developed in order to realize a multi-channel sound system wherein appropriate output (balance and fader) control can be conducted without adversely affecting the program contents, and a sound system, which can reproduce a plurality of channel signals including at least left and right front channels and a center channel for forward-placed speakers, comprises a gain controller to control the gain of either left or right channel signal and a gain controller to control the gain of the center channel signal depending on the gain control of the left or right channel signal.

Abstract: The invention relates to an audio reproducing apparatus having a converting means for converting main audio signals included in an audio signal for a plurality of channels into audio signals that provide a reproduction sound field similar to that obtained in reproduction by means of speakers even when the audio signals are reproduced by means of headphones; and an adding means for adding an auxiliary audio signal included in the audio signal for a plurality of channels to an audio signal outputted from the converting means; wherein an audio signal outputted from the adding means is supplied to headphones.

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: Methods and apparatus for recording and playing back multi-channel digital audio having different sampling rates for different channels are provided. Recording is accomplished by sampling a plurality of channels of audio data and arranging the samples of the plurality of channels of audio data in most nearly the order needed during playback. Playback is accomplished by accessing the samples of the plurality of channels of audio data and converting the samples from digital to analog form. The playback may include deriving audio signals in a different format from the format in which the programming is recorded.

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: When a first bi-wiring speaker is connected to output terminals 5A and 5B and a second bi-wiring speaker is connected to output terminals 5D and 5E, and further a two-channel stereo reproduction is performed, a signal 3L for a L channel outputted from a decoder 2 is amplified by amplifiers 4A and 4B in parallel to be given to the first bi-wiring speaker in a bi-wiring connecting manner and a signal 3R for a R channel outputted from the decoder 2 is amplified by amplifiers 4C and 4D in parallel and given to the second bi-wiring speaker in a bi-wiring connecting manner. At this time, a gain corrector 7 equalizes a gain which an output signal of the amplifier 4A has with respect to the signal 3L for the L channel with a gain which an output signal of the amplifier 4B has with respect to the signal 3L for the L channel.

Abstract: In digital audio decoding, the compressed data of the first two channels of a frame of MPEG multichannel audio are received, parsed and stored in plural spaced sub-blocks of buffer memory. Then, coding information of the same frame is received and interpreted. If the signal includes MPEG-2 multichannel audio, then compressed data of the remaining channels of the frame are received, parsed and interleaved in the sub-blocks. The sub-blocks are read from the buffer and decoded, sub-block by sub-block, transformed into decoded output data of all of the channels for simultaneous output to output devices. Less than 8K-bits read to local memory can be decoded to produce full multiple channel output. A memory controller accesses non-contiguous areas of the buffer memory and reads and writes differing sizes of data blocks. Substantial savings in both on and off chip memory are provided.

Abstract: The ratio y(n) of two digital values, respectively a(n) and b(n), representing the n.sup.th elements of two respective sequences of digital input data representing two quantities slowly varying in time, is obtained by computingy(n)=y(n-1)+g*[a(n)-b(n)*y(n-1)]wherein g represents a multiplying factor. Within the domain of the z transform, the expression becomes:Y(z)=z.sup.-1 *Y(z)+g[A(z)-B(z)conv Y(z)*z.sup.-1where conv indicates an operation of convolution and which, for input sequences corresponding to signals filtered through a lowpass filter with a time constant greater than or equal to 3 msec is simplified to: ##EQU1## The approximation is exceptionally good and computation thereof may be achieved by the use of relatively simple hardware, without severely burdening the workload of a microprocessor.

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: An audio enhancement system and method for use in a surround sound environment creates a more diffuse and continuous sound field from a multi-channel, multi-speaker reproduction environment. Multiple audio source signals generated from an audio recording, which are intended for speakers placed in front of and behind a listener, are isolated into pairs and processed to create corresponding pairs of component audio signals. Each pair of component audio signals is generated, at least in part, from the information present in both corresponding audio source signals. The individual component audio signals are then selectively combined to form enhanced output signals so that each enhanced output signal is modified as function of a plurality of audio source signals.

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 stereophonic signal processing system for enhancing the perception of sonic dimension and imaging in a sound reproduction system having at least two stereo speakers.

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 digital surround sound decoder. The decoder uses an architecture including two signal processing chips to achieve a program that can decode audio data at sufficiently high resolution. The decoder performs in real time and is compatible with standard surround sound formats. The decoder includes software that utilizes table lookups for critical functions in the decoding process. The processing flow of the decoder's program takes advantage of the multi-function capability within the specific processors used in the design while using a minimum number of program instructions. The program implements band pass filtering, sum-difference calculations, fast-attack slow-decay integration, summation and reciprocal processing, determination of fast and slow modes, look-up table indexing, adaptive matrix processing and various other functions to generate decoded surround sound signals from encoded left and right signal inputs.

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 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: An audio amplifier arrangement contains reproduction channels (R, L, C, SR) with corresponding channel outputs, to which a selectable number and/or assignment of different combinations of sound radiators can be connected in the form of loudspeakers (Lr, Ll) and/or external loudspeaker boxes (Br, Bl, Bc) to obtain different reproduction modes, and a signal processor which can be switched in such a way, that different audio signals (Vr', Vl', Vc, Vcr) are made available for the reproduction channels, to obtain the different reproduction modes. Connecting the corresponding external loudspeaker boxes to predetermined channel outputs requires a change in the reproduction mode, whereby the assignment of already connected sound radiators is changed, and/or the signal processor is switched. According to the invention, part of the channel outputs have first and second switching means (Sr, Sl, Sc), which automatically make the required change in the operating mode when external loudspeaker boxes are connected.

Abstract: A matrix decoding system and method generates left, right, center and surround output signals by adding and subtracting weighted versions of first and second input signals. The output signals are generated using a variable matrix to combine the input signals. The variable matrix is controlled by steering control signals derived from a controller which generates the steering control signals responsive to the detected phase difference between the first and second input signals. The decoding system and method uses compressors to generate constant amplitude versions of the input channels prior to performing phase detection. Because the compressors generate constant amplitude signals, phase detection is not affected by an imbalance in channel power.

Abstract: A frequency-dependent linear audio signal processor takes source signals S in input signals and provide directionally spread directionally encoded output signals. The processor directionally encodes with constant gain magnitude frequency components of the source signal S to-and-fro across a predetermined directional stage P" as frequency increases such that at least three predetermined positions within the stage P", the directional encoding has substantially zero perceived phasiness. The processor may be a frequency-dependent rotation matrix for stereo input signal and may be a unitary network using a feedback path around parallel identical all-pass networks in series with a rotation matrix and a feedforward path bypassing the all-pass networks. Successive frequencies of positioning of source signal S at a predetermined position P within the stage P" are preferably spaced approximately uniformly on a logarithmic or Bark Frequency scale.

Abstract: A surround sound processor system for multichannel redistribution of stereophonic signals has digitally controlled gains in each input and each output channel, controlled by a microprocessor, which receives an input signal from a microphone placed at the preferred listening location within the listening area for automatically balancing the input signals and setting both input and output gains during a calibration process so as to provide the listener with the best possible surround sound reproduction of the stereophonic source material. As a visual aid, the microprocessor displays menus and messages on a video screen, and a visual display shows the relative levels of the six axes of control signals within the surround sound processor.

Abstract: A surround sound processor for presenting stereophonic audio signals on a number of loudspeakers surrounding a listening area, comprising an input matrix stage, a detector filter and matrix circuit, a direction detector circuit incorporating improved filters, a novel detector splitter circuit providing three direction signals from the two direction detector circuit outputs, a novel three-channel servologic circuit with improved variable filters responding selectively to the rates of change of the direction signals and providing six control voltage signals through linearity correction networks to six voltage-controlled amplifiers, the outputs of which are combined in an output matrix to provide a number of loudspeaker feed signals through buffer amplifiers to the output terminals of the processor. The detector splitter circuit is configurable to either a forward oriented or a backward oriented mode of operation, corresponding changes being made to the voltage-controlled amplifiers and output matrix circuitry.