Abstract: A directional signal processing system for beamforming information signals. The system includes an oversampled filterbank, which has an analysis filterbank for transforming the information signals in time domain into channel signals in transform domain, a synthesis filterbank and a signal processor. The signal processor processes the outputs of the analysis filterbank for beamforming the information signals. The synthesis filterbank transforms the outputs of the signal processor to a single information signal in time domain.

Abstract: An adaptive signal processing system for improving a quality of a signal. The system includes an analysis filterbank for transforming a primary information signal in time domain into oversampled sub-band primary signals in frequency domain and an analysis filterbank for transforming a reference signal in time domain into oversampled sub-band reference signals. Sub-band processing circuits process the signals output from the filterbanks to improve a quality of an output signal. A synthesis filterbank can combine the outputs of the sub-band processing circuits to generate the output signal.

Abstract: A system and method of improving signal intelligibility over an interference signal is provided. The system includes a psychoacoustic professor having a psychoacoustic model wherein the level of a signal-of-interest is improved so as to be audible above noise and so as not to exceed a predetermined maximum output level. The system can be combined with active noise cancellation.

Abstract: Method and system for protecting content in a programmable system is provided. The system is connectable to an external device through one or more access ports. Content protection method/manager is implemented by assigning one or more access modes to the access port and switching the access modes. In response to a current access mode, the content protection method/manager restricts visibility of the system to the external device via the access port.

Abstract: A method and system for reducing audible side effects of dynamic current consumption is provided. The system includes an audio subsystem, and a plurality of digital subsystems. The audio subsystem and the digital subsystem are powered by a common power supply. The digital subsystem processes data packets including audio data packets. The processing events implemented in one or more than one digital subsystem are reorganized to change the profile for executing the events inside the subsystem(s). The dynamic current spectral properties in one or more digital subsystems are changed.

Abstract: A method and system for data logging in a listening device is provided. The system includes a digital signal processing (DSP) entity, which performs normal hearing aid audio and system processing, a level translating module, and a non-volatile (NV) memory. The NV memory is used to store logged data. During the hearing aid audio processing, the DSP entity communicates with the NV memory via the level translating module. The level translating module performs voltage-translation during data logging to a communication signal between the DSP entity and the NV memory.

Abstract: A lighted sign for application to a vehicle surface includes first and second metallic layers with a magnetic plate sandwiched therebetween. The second metallic layer includes a circumference smaller than that of the magnetic plate such that it is pressed against a vehicle surface when the magnetic plate is magnetically adhered thereto. A light source is coupled to the magnetic plate with a positive wire connected to the first metallic layer and a negative wire connected to the second layer such that the light source emits light when the first metallic layer is energized by a vehicle electrical power source and is grounded when the second metallic layer is pressed against the vehicle metallic surface. The magnetic plate and first metallic surface include cutouts so that light may be emitted by the light source therethrough. The lighted sign 10 include a microcontroller and environmental sensors for advanced lighting control.

Abstract: An acoustic shock protection method and device are provided. A pattern analysis-based approach is taken to an input signal to perform feature extraction. A parameter space is identified, which is corresponding to the signal space of the input signal. A rule-based decision approach is taken to the parameter space to detect an acoustic shock event. The device may be advantageously implemented using a weighted overlap-add approach to provide low group delay, high-fidelity and a high degree of protection from acoustic shock events.

Abstract: A sound intelligibility enhancement (SIE) system is disclosed. The SIE system uses a psychoacoustic model and preferably an oversampled filterbank wherein the level of a signal-of-interest that falls below the environmental noise is selectively amplified as a function of the input level and frequency so that it is audible above the noise but never exceeds a predetermined maximum output level as a function of frequency. The SIE system can be combined with active noise cancellation.

Abstract: A filterbank structure is provided which provides a flexible compromise between the conflicting goals of processing delay, filter sharpness, memory usage and band interaction. The filterbank has an adjustable number of bands and a stacking which provides for a selectable shift of band frequencies to one of two discrete sets of center frequencies. The width of the bands and hence the number of the bands is selected depending upon acceptable delay, memory usage, and processing speed required. The flexibility in terms of stacking of the bands provides twice the number of potential band edge placements, which is advantageous for hearing loss fitting especially at low frequencies. The same filter coefficients can be used for analysis and synthesis, to reduce memory usage.

Abstract: An adaptive signal processing system for improving a quality of a signal. The system includes an analysis filterbank for transforming a primary information signal in time domain into oversampled sub-band primary signals in frequency domain and an analysis filterbank for transforming a reference signal in time domain into oversampled sub-band reference signals. Sub-band processing circuits process the signals output from the filterbanks to improve a quality of an output signal. A synthesis filterbank can combine the outputs of the sub-band processing circuits to generate the output signal.

Abstract: A directional signal processing system for beamforming information signals. The system includes an oversampled filterbank, which has an analysis filterbank for transforming the information signals in time domain into channel signals in transform domain, a synthesis filterbank and a signal processor. The signal processor processes the outputs of the analysis filterbank for beamforming the information signals. The synthesis filterbank transforms the outputs of the signal processor to a single information signal in time domain.

Abstract: A method for equalizing output signals from a plurality of signal paths is disclosed. The method comprises steps of identifying a transfer function for each of signal paths, determining a filtering function for each signal path such that a product of the transfer function, and the filtering function is a selected function and applying the filtering function to the corresponding signal path, thereby correcting the transfer function of the signal path to the selected function to equalize the output signals from the signal paths. The step of applying the filtering function comprises steps of providing an equalization filter to the signal path and applying the filtering function to the equalization filter of its corresponding signal path, thereby equalizing output signals from the filter of the signal paths.

Abstract: A filterbank structure is provided which provides a flexible compromise between the conflicting goals of processing delay, filter sharpness, memory usage and band interaction. The filterbank has an adjustable number of bands and a stacking which provides for a selectable shift of band frequencies to one of two discrete sets of center frequencies. The width of the bands and hence the number of the bands is selected depending upon acceptable delay, memory usage, and processing speed required. The flexibility in terms of stacking of the bands provides twice the number of potential band edge placements, which is advantageous for hearing loss fitting especially at low frequencies. The same filter coefficients can be used for analysis and synthesis, to reduce memory usage.

Abstract: There is provided a method of controlling feedback in an acoustic system, for example a digital hearing aid, in which there is a potential feedback path between the output and the input. The method comprises making a spectral estimate of the input signal spectrum, and then subjecting the spectral estimate to a psycho-acoustic model to generate a control signal. A noise source is passed through a shaping filter, which is controlled with the control signal, to generate frequency-shaped noise, which is inaudible to someone hearing the output. The frequency-shaped noise is then added to the input signal to form a combined signal, which is processed in a forward path, to generate a first output signal. The first output signal and the frequency-shaped noise signal are analyzed, to determine the presence of feedback at difference frequencies, and the characteristics of the forward path are modified to reduce the gain thereof at frequencies where feedback is detected.

Abstract: An apparatus for a hearing aid provides an application specific integrated circuit (ASIC) for filtering of input signals and a programmable digital signal processor connected to it, for control of filterbank gains. This provides a compromise between the conflicting goals of size, re-programmability and power consumption. The fixed portion of the processing, i.e. filtering is implemented in hardware in the ASIC, and the variable or adjustable portion of the processing is implemented by the programmable digital signal processor. The filterbank has an adjustable number of channels and means for shifting the center frequencies of the bands in unison to one of two discrete sets of center frequencies. A wide range of hearing loss compensation schemes can be implemented. Software programs can be executed on the programmable digital signal processor.

Abstract: A filterbank structure is provided which provides a flexible compromise between the conflicting goals of processing delay, filter sharpness, memory usage and band interaction. The filterbank has an adjustable number of bands and a stacking which provides for a selectable shift of band frequencies to one of two discrete sets of center frequencies. The width of the bands and hence the number of the bands is selected depending upon acceptable delay, memory usage, and processing speed required. The flexibility in terms of stacking of the bands provides twice the number of potential band edge placements, which is advantageous for hearing loss fitting, especially at low frequencies. The same filter coefficients can be used for analysis and synthesis, to reduce memory usage.

Abstract: A method is provided for programming a digital hearing aid using a program encoded in an audio band (20 Hz-20 kHz) signal, to transmit and verify programs and algorithm parameters. Preferably, this is in a digital hearing aid including filterbanks, filtering the audio signal into different frequency bands. The signal is encoded by the presence and absence of a signal in each frequency band or by other well-known modulation techniques used by computer modems. Special programming signals are provided alternating between the frequency bands in a manner to clearly distinguish the program data from any other interfering or normally present audio signal. The method does not require additional hardware, and offers reduced power consumption, as compared to some known wireless programming interfaces. It enables remote programming over a network using standard multimedia computer hardware.