Abstract: A method of automated feedforward filter design comprising designing a feedforward filter for a system implementing active noise cancelling is described. The method includes designing the feedforward filter by determining a filter transfer function of the feedforward filter. The filter transfer function is determined using a least square method. The method also includes determining the filter transfer function by defining a target transfer function of the feedforward filter and applying the least square method using the target transfer function to determine a filter expression for the filter transfer function. The least square method is a weighted least square method.

Abstract: A computer-implemented method for automatically designing a feedforward filter that is optimized for an audio transparency mode of operation of an audio device, the audio device comprising the feedforward filter and a feedback filter, the method comprising optimizing the feedforward filter for the audio transparency mode of operation, wherein optimization of the feedforward filter is dependent on the feedback filter.

Abstract: A key word detection apparatus and a method for low-power voice-activated devices are presented. A first signal processing module operates with a first transducer to receive an incoming signal and generates a first sample. A second signal processing module operates with a second transducer which receives an incoming signal and generates a second sample. In summary, a signal processing system, in particular a key word detection system, has a first low power module that wakes up a second higher power module. The second module uses signals from the first module in order to improve accuracy of key word detection or other signal processing tasks.

Abstract: A digital signal processing system for multiplying a digital value and a digital signal. The digital signal processing system receives the digital value in an encoded format, and multiplies the digital value with the digital signal. The digital value in the encoded format has an offset, which is encoded as a floating point. The disclosure provides a digital processing system that can carry out a multiplication operation with a smaller area, less complexity and/or reduced power usage compared with known multipliers.

Abstract: A digital filter for filtering a pulse density modulation (PDM) signal is presented. The filter has a first filter circuit to receive an input signal and to provide a filtered input signal at successive time steps which include a first filtered value at the first time step and a second filtered value at a second time step. The filter also has a quantizer to provide an output signal comprising output values at successive time steps and a filter variable circuit with a first multiplication circuit to receive the first filter variable, and divide the first filter variable by a first gain factor and a first summing circuit configured to receive the divided first filter variable, receive the output signal, and add the divided first filter variable and the first output value and a second multiplication circuit and a delay circuit.

Abstract: A computer-implemented method for automatically optimizing a hybrid active noise cancellation system, the hybrid active noise cancellation system comprising a feedback filter and a feedforward filter, the method comprising optimizing the feedforward filter, thereby optimizing the hybrid active noise cancellation system, wherein optimization of the feedforward filter is dependent on the feedback filter.

Abstract: A method of automated feedforward filter design comprising designing a feedforward filter for a system implementing active noise cancelling is described. The method includes designing the feedforward filter by determining a filter transfer function of the feedforward filter. Optionally, the filter transfer function is determined using a least square method. The method also includes determining the filter transfer function by defining a target transfer function of the feedforward filter and applying the least square method using the target transfer function to determine a filter expression for the filter transfer function. Optionally, the least square method is a weighted least square method.

Abstract: A digital filter structure and related method of digital filtering are presented. The digital filter structure is arranged to receive one or more clocked input signals having a first clock rate, and which is driven at a second clock rate higher than said first clock rate. The digital filter structure has a plurality of delay elements and multiplexing circuitry arranged to selectively engage the delay elements such that, at every clock cycle of the digital filter structure, a filter operation is performed on a different stream of data. The disclosure can be applied in many different contexts. One particular implementation example is that of an adaptive noise cancellation (ANC) system using sigma-delta infinite impulse response filters. In this context the present disclosure minimizes latency and hardware implementation area by requiring only one filtering circuit for multiple channels of data to be filtered.

Abstract: An anti-noise signal generator and a method of generating an anti-noise signal are presented. The anti-noise generator includes a first microphone input to receive a first sigma-delta modulated signal at a microphone sampling frequency. The first microphone input is coupled to a combiner via a first path and a second path. The combiner is adapted to combine a first filtered signal from the first path and a second filtered signal from the second path to generate the anti-noise signal. The first path includes a first digital filter adapted to operate at a filter frequency equal or greater than the microphone sampling frequency. The second path includes a second digital filter. The first digital filter may be a sigma-delta based filter that includes a sigma-delta modulator.

Abstract: An anti-noise signal generator and a method of generating an anti-noise signal are presented. The anti-noise generator includes a first microphone input to receive a first sigma-delta modulated signal at a microphone sampling frequency. The first microphone input is coupled to a combiner via a first path and a second path. The combiner is adapted to combine a first filtered signal from the first path and a second filtered signal from the second path to generate the anti-noise signal. The first path includes a first digital filter adapted to operate at a filter frequency equal or greater than the microphone sampling frequency. The second path includes a second digital filter. The first digital filter may be a sigma-delta based filter that includes a sigma-delta modulator.

Abstract: A signal processing structure and method are presented. A first digital filter operates on received sigma-delta modulated (SDM) input signals. A second pre-processing digital filter receives a SDM input signal, directly low pass filter the SDM input signal and provides an output SDM signal. The output sigma-delta modulated signal is provided as an input for said first digital filter. In standard digital systems operating with digital microphones, filtering of the microphones' output signal requires to first convert the signal into pulse code modulation (PCM), then filter and finally convert back to pulse density modulation (PDM). This approach increases the latency of the system because decimation and interpolation must be performed in order to pass from PDM to PCM. By using filters that operate directly on the oversampled PDM output of the digital microphones it is possible to reduce the latency of the system and minimize the hardware area.

Abstract: A digital filter and a method for filtering a pulse density modulation (PDM) signal are presented. The digital filter has a first filter circuit to receive an input signal with input values at successive time steps to provide a filtered input signal with filtered values at successive time steps. The digital filter does not require sample-rate or data format conversions. Also, the digital filter is area and power efficient when implemented in hardware. Optionally, the digital filter has a sigma-delta modulator including the quantiser, the sigma-delta modulator being used to receive the filtered input signal and to process the filtered input signal before and/or after being quantised by the quantiser. This digital filter does not require sample-rate or data format conversions. This digital filter is area and power efficient when implemented in hardware.

Abstract: A quantizer and a method for a sigma-delta modulator circuit that may be used as a component within an adaptive-noise cancelling headphone are presented. An apparatus includes a quantizer to receive an input signal with successive input values and quantizes the input signal at discrete intervals. This is done by mapping the input value of the input signal at each interval to one of a plurality of quantization levels with three or more quantization levels that are non-uniformly spaced. The plurality of quantization levels has a first portion with two or more quantization levels having the same sign and being proportional to a first fraction having one as its numerator and two to a power of a first variable as its denominator, the first variable being an integer and having a different value for each of the two or more quantization levels of the first portion.

Abstract: A digital filter and a method for filtering a pulse density modulation (PDM) signal are presented. The digital filter has a first filter circuit to receive an input signal with input values at successive time steps to provide a filtered input signal with filtered values at successive time steps. The digital filter does not require sample-rate or data format conversions. Also, the digital filter is area and power efficient when implemented in hardware. Optionally, the digital filter has a sigma-delta modulator including the quantiser, the sigma-delta modulator being used to receive the filtered input signal and to process the filtered input signal before and/or after being quantised by the quantiser. This digital filter does not require sample-rate or data format conversions. This digital filter is area and power efficient when implemented in hardware.

Abstract: An anti-noise signal generator and a method of generating an anti-noise signal are presented. The anti-noise generator includes a first microphone input to receive a first sigma-delta modulated signal at a microphone sampling frequency. The first microphone input is coupled to a combiner via a first path and a second path. The combiner is adapted to combine a first filtered signal from the first path and a second filtered signal from the second path to generate the anti-noise signal. The first path includes a first digital filter adapted to operate at a filter frequency equal or greater than the microphone sampling frequency. The second path includes a second digital filter. The first digital filter may be a sigma-delta based filter that includes a sigma-delta modulator.

Abstract: A signal processor and a method for processing an input signal are presented. The signal processor is adapted to clip an oversampled input signal without introducing noise in the frequency band of interest. For instance, the signal processor may be used for clipping an acoustic signal. The signal processor includes a summer coupled to a limiter and to a feedback circuit. The summer is adapted to sum the input signal with at least one feedback signal to provide an adjusted signal. The limiter is adapted to compare the adjusted signal with a first threshold value and a second threshold value to provide a limited signal. The feedback circuit is adapted to calculate a difference between the limited signal and the adjusted signal, and to generate at least one feedback signal based on the difference.

Abstract: A sensor fusion method of calculating an orientation of an object by combining readings from different types of orientation sensors to estimate the orientation of an object. An analytical solution is provided which is computationally efficient and can be implemented in fixed or floating point architecture. The method comprises receiving an input orientation; receiving a reading from a first orientation sensor; receiving a reading from a second orientation sensor; where said first and second orientation sensors are of different types; and determining an updated orientation by calculating a rotation based on the orientation sensor readings and applying the calculated rotation to the input orientation.

Abstract: Method and apparatus for controlling a digitized analog audio device from a control circuit communicating with the digitized analog audio device with a serial digital analog audio protocol on a serial communication medium multiplexes command data words within digitized analog audio data frames. The digitized analog audio device extracts the command data words from the digitized analog audio data frames. The command data word includes a keyword packet, a command packet, an optional address packet, and an optional data packet. The keyword packet, a command packet, an optional address packet, and an optional data packet are each inserted into a separate number of truncated digitized analog audio data frames for iterative and successive transmission. The iterative transmission of the packets of the command word decreases the likelihood that digitized analog audio frames would contain packets of the command word.

Abstract: An Encryption processor is disclosed comprising an S-box unit, a byte permutation unit, a MixColumns unit and a key expansion unit. In an embodiment the byte permutation unit comprises a data input terminal for receiving input data, two or more row processors, each row processor having an input coupled to the input terminal and an output, and a clock gating circuit for generating for each of the two or more row processors a gated clock signal. Furthermore, a clock gated key expansion unit is disclosed. By clock gating, the power consumption of the processor is reduced which extends the battery life time in battery powered devices.

Abstract: Method and apparatus for controlling a digitized analog audio device from a control circuit communicating with the digitized analog audio device with a serial digital analog audio protocol on a serial communication medium multiplexes command data words within digitized analog audio data frames. The digitized analog audio device extracts the command data words from the digitized analog audio data frames. The command data word includes a keyword packet, a command packet, an optional address packet, and an optional data packet. The keyword packet, a command packet, an optional address packet, and an optional data packet are each inserted into a separate number of truncated digitized analog audio data frames for iterative and successive transmission. The iterative transmission of the packets of the command word decreases the likelihood that digitized analog audio frames would contain packets of the command word.