Abstract: In a method of lossless processing of an integer value signal in a prediction filter which includes a quantiser, a numerator of the prediction filter is implemented prior to the quantiser and a denominator of the prediction filter is implemented recursively around the quantiser to reduce the peak data rate of an output signal. In the lossless processor, at each sample instant, an input to the quantiser is jointly responsive to a first sample value of a signal input to the prediction filter, a second sample value of a signal input to the prediction filter at a previous sample instant, and an output value of the quantiser at a previous sample incident. In a preferred embodiment, the prediction filter includes noise shaping for affecting the output of the quantiser.

Abstract: In a method of lossless processing of an integer value signal in a prediction filter which includes a quantiser, a numerator of the prediction filter is implemented prior to the quantiser and a denominator of the prediction filter is implemented recursively around the quantiser to reduce the peak data rate of an output signal. In the lossless processor, at each sample instant, an input to the quantiser is jointly responsive to a first sample value of a signal input to the prediction filter, a second sample value of a signal input to the prediction filter at a previous sample instant, and an output value of the quantiser at a previous sample incident. In a preferred embodiment, the prediction filter includes noise shaping for affecting the output of the quantiser.

Abstract: In a method of lossless processing of an integer value signal in a prediction filter which includes a quantiser, a numerator of the prediction filter is implemented prior to the quantiser and a denominator of the prediction filter is implemented recursively around the quantiser to reduce the peak data rate of an output signal. In the lossless processor, at each sample instant, an input to the quantiser is jointly responsive to a first sample value of a signal input to the prediction filter, a second sample value of a signal input to the prediction filter at a previous sample instant, and an output value of the quantiser at a previous sample incident. In a preferred embodiment, the prediction filter includes noise shaping for affecting the output of the quantiser.

Abstract: In a method of lossless processing of an integer value signal in a prediction filter which includes a quantiser, a numerator of the prediction filter is implemented prior to the quantiser and a denominator of the prediction filter is implemented recursively around the quantiser to reduce the peak data rate of an output signal. In the lossless processor, at each sample instant, an input to the quantiser is jointly responsive to a first sample value of a signal input to the prediction filter, a second sample value of a signal input to the prediction filter at a previous sample instant, and an output value of the quantiser at a previous sample incident. In a preferred embodiment, the prediction filter includes noise shaping for affecting the output of the quantiser.

Abstract: In a method of lossless processing of an integer value signal in a prediction filter which includes a quantiser, a numerator of the prediction filter is implemented prior to the quantiser and a denominator of the prediction filter is implemented recursively around the quantiser to reduce the peak data rate of an output signal. In the lossless processor, at each sample instant, an input to the quantiser is jointly responsive to a first sample value of a signal input to the prediction filter, a second sample value of a signal input to the prediction filter at a previous sample instant, and an output value of the quantiser at a previous sample incident. In a preferred embodiment, the prediction filter includes noise shaping for affecting the output of the quantiser.

Abstract: A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6).

Abstract: A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6).

Abstract: Matrix reproduction decoding means derive from input signals intended to feed a stereophonic plurality of loudspeakers output signals intended to feed a second greater plurality of loudspeakers in a stereophonic arrangement covering a sector of directions, substantially so as to preserve total reproduced energy to within an overall gain and equalization, and to preserve to within constants of proportionality the angular dispositions of reproduced acoustical velocity and sound intensity vectors at an ideal listening position. Preferably for two-channel signals matrix means is frequency-dependent giving increased angular width above 5 kHz, and may incorporate width control. Matrix means encoding loudspeaker feed signals into transmission channel signals, and matrix means decoding transmission channel signals into loudspeaker feed signals may be used giving overall matrix means in accordance with the invention.

Abstract: An audio signal processing system produces an output 24 having an illusory distance effect for a sound source signal S by feeding it via a direct signal path 25 and an indirect signal path 22, 23 passing through early reflection simulation apparatus 1 which feed an output mixing mechanism 9. A control system adjusts the relative delays 3, 4 and relative gains 5, 6 in the direct 25 and indirect 22, 23 signal paths to modify the illusory distance effect so as to substantially maintain the mathematical relationship between the gains and time delays of simulated reflections relative to first sound arrivals at the output 24 encountered for sounds at that source distance in actual rooms. Signal paths 22, 23, 24, 25 may be stereophonic or multichannel using matrix gain coefficients in the early reflection simulator 1, and may produce different simulated distances for different sound positions. A plurality of sound sources S may have different simulated distances while feeding a common early reflection simulator 1.

Abstract: A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6).

Abstract: A decoder is provided for feeding an irregular array of m (being three or more) pairs of diametrically opposite loudspeakers, each loudspeaker being disposed at an equal distance r from a common reference point. The decoder incorporates a WXY circuit 10 for producing output signals W, X, Y and -jW from the input signals, and shelf filters 12, 14, 16 and 22 and high-pass filters 18, 20 and 24 for producing output signals W', X', Y' and -jW.sub.i.sup.". In addition the decoder includes an amplitude matrix circuit 26 for producing signals S.sub.i.sup.+ and S.sub.i.sup.-, to be fed to the loudspeakers of each pair, which satisfy particular gain requirements, whereby the outputs of the loudspeakers are adapted to irregular positioning of the loudspeakers which may be dictated by room geometry. A decoder is also provided for feeding a three-dimensional loudspeaker layout.

Abstract: A basic decoder designed to feed signals to at least three loudspeakers disposed at respective azimuths around a reference point at equal distances therefrom is modified to make it suitable for feeding loudspeakers at the same azimuths but at non-uniform distances from the reference point. Each output of the basic decoder is subject to a time delay proportional to the difference of time of travel of sound from the first pair of loudspeakers to the reference point and the time of travel of sound from the second pair of loudspeakers to the reference point, divided by the sum of said times of travel to an adder connected between the first velocity signal input and the corresponding high-pass filter.

Abstract: In a surround sound reproduction system the required number of power amplifiers is reduced by effecting at least part of the decoding operation by the manner in which the various loudspeakers are connected to the outputs of the power amplifiers while preserving linear independence of the signals fed to every set of three loudspeakers. In one embodiment, the loudspeakers are grouped in pairs such that the sum of signals fed to the loudspeakers of each pair is the same for all of the pairs, a first amplifier is arranged to receive said signal and a respective second amplifier for each pair of loudspeakers is arranged to receive the signal to be fed to one of the loudspeakers of such pair, the second loudspeaker of each pair being connected between the output of the corresponding second amplifier and the output of the first amplifier.