Abstract: A sound capture device comprises a symmetric microphone array that includes non-radially-oriented directional sensors (101). The device typically derives a spherical harmonic representation of the incident sound field, and affords higher signal-to-noise ratios and better directional fidelity than prior arrays, across a wide range of audio frequencies.

Abstract: A low delay corrector (LDC) unit includes a non-linear function generator and a filter. The nonlinear function generator receives a first signal and outputs a second signal in dependence on the first signal and a transfer function of the nonlinear function generator. The filter is fed in dependence on the second signal output by the nonlinear function generator. The first signal received by the nonlinear function generator is derived in dependence on an input signal provided to an input of the LDC unit and an output of the filter. An output of the LDC unit is derived in dependence on the first signal received by the nonlinear function generator and the second signal output by the nonlinear function generator.

Abstract: A sound capture device comprises a symmetric microphone array that includes non-radially-oriented directional sensors (101). The device typically derives a spherical harmonic representation of the incident sound field, and affords higher signal-to-noise ratios and better directional fidelity than prior arrays, across a wide range of audio frequencies.

Abstract: Systems and methods for performance improvements in digital switching amplifiers using low-pass filtering to reduce noise and distortion. In one embodiment, a digital pulse width modulation (PWM) amplifier includes a signal processing plant configured to receive and process an input audio signal. The amplifier also includes a low-pass filter configured to filter audio signals output by the plant. The filtered output of the plant is added to the input audio signal as feedback. The plant may consist of a modulator and power switch, a noise shaper, or any other type of plant. An analog-to-digital converter (ADC) may be provided to convert the output audio signal to a digital signal. Filtering may be implemented before or after the ADC, and a decimator may be placed after the ADC if it is an oversampling ADC.

Abstract: Systems and methods in which an ultrasonic signal is introduced into an audio signal before the audio signal is amplified by a switching amplifier. The added ultrasonic signal (e.g., a tone at half the amplifier's switching frequency) shifts the signals input to a set of power switches so that they do not switch nearly simultaneously. The ultrasonic signal causes the output current to be well defined to eliminate dead time distortion at low signal levels. Adding the tone ultrasonic signal causes the distortion to shift to an amplitude greater than zero. Signals that exceed this amplitude will experience the distortion, but the distortion will be less noticeable than in lower-amplitude signals. Signals that do not exceed this amplitude will not experience the distortion at all. Adding an ultrasonic signal may also draw energy away from the switch frequency and its harmonics to interference with AM radio reception.

Abstract: Systems and methods for improving the stability of feedback and/or feed-forward subsystems in digital amplifiers. One embodiment comprises a digital pulse width modulation (PWM) controller. The controller includes an input for receiving a digital audio input signal and is configured to generate a PWM output signal based on the input signal at an output. The controller also has control inputs for receiving external audio correction signals such as feedback and power supply feed-forward signals. The controller has correction circuitry for processing the received external control signals and modifying the input signal based on these signals. Fault detectors monitor fault conditions at various locations within the correction circuitry, and a protection control unit receives fault signals from the fault detectors and modifies operation of the controller in response to the fault signals.

Abstract: Systems and methods for improving the stability of feedback and/or feed-forward subsystems in digital amplifiers. One embodiment comprises a digital pulse width modulation (PWM) controller. The controller includes an input for receiving a digital audio input signal and is configured to generate a PWM output signal based on the input signal at an output. The controller also has control inputs for receiving external audio correction signals such as feedback and power supply feed-forward signals. The controller has correction circuitry for processing the received external control signals and modifying the input signal based on these signals. Fault detectors monitor fault conditions at various locations within the correction circuitry, and a protection control unit receives fault signals from the fault detectors and modifies operation of the controller in response to the fault signals.

Abstract: Systems and methods in which an ultrasonic signal is introduced into an audio signal before the audio signal is amplified by a switching amplifier. The added ultrasonic signal (e.g., a tone at half the amplifier's switching frequency) shifts the signals input to a set of power switches so that they do not switch nearly simultaneously. The ultrasonic signal causes the output current to be well defined to eliminate dead time distortion at low signal levels. Adding the tone ultrasonic signal causes the distortion to shift to an amplitude greater than zero. Signals that exceed this amplitude will experience the distortion, but the distortion will be less noticeable than in lower-amplitude signals. Signals that do not exceed this amplitude will not experience the distortion at all. Adding an ultrasonic signal may also draw energy away from the switch frequency and its harmonics to interference with AM radio reception.

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: An encoder for a data compression system comprises a core encoder and a FIFO buffer. The FIFO buffer is able to absorb short bursts of high data rate and deliver the data to a transmission medium at a rate that it can handle. A corresponding decoder includes a FIFO buffer which receives data at a rate not exceeding the maximum that the transmission medium can deliver, but delivers data to a core encoder at a rate which can, for short periods, greatly exceed the rate from a transmission medium. With consumer media such as CD and DVD, the decoder cost is paramount so it is important to minimize the size of buffer memory required. Another consideration is the delay caused by the buffering since it is important to the latency between the user selecting a track or index point and hearing the decoded audio.

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: A signal convertor comprising a pulse modulator, and a modifier for modifying the signal input thereto in dependence upon the error in previous values of the output thereof, to reduce the effects of said error within a desired signal band.

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).