Abstract: A computer-implemented method including receiving a first signal from an input device of a hearing aid. The first signal may include a noise signal. The computer-implemented method may include low-pass filtering first periodic samples of the first signal, and the first periodic samples may be approximately periodic with respect to a period of the noise signal. The computer-implemented method may further include low-pass filtering second periodic samples of the first signal, and the second periodic samples may be approximately periodic with respect to the period of the noise signal. The second periodic samples may also be phase shifted relative to the first periodic samples. Hearing aid systems and apparatuses are also disclosed.

Abstract: A computer-implemented method including receiving a first signal from an input device of a hearing aid. The first signal may include a noise signal. The computer-implemented method may include low-pass filtering first periodic samples of the first signal, and the first periodic samples may be approximately periodic with respect to a period of the noise signal. The computer-implemented method may further include low-pass filtering second periodic samples of the first signal, and the second periodic samples may be approximately periodic with respect to the period of the noise signal. The second periodic samples may also be phase shifted relative to the first periodic samples. Hearing aid systems and apparatuses are also disclosed.

Abstract: A computer-implemented method including receiving a first signal from an input device of a hearing aid. The first signal may include a noise signal. The computer-implemented method may include low-pass filtering first periodic samples of the first signal, and the first periodic samples may be approximately periodic with respect to a period of the noise signal. The computer-implemented method may further include low-pass filtering second periodic samples of the first signal, and the second periodic samples may be approximately periodic with respect to the period of the noise signal. The second periodic samples may also be phase shifted relative to the first periodic samples. Hearing aid systems and apparatuses are also disclosed.

Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: A simplified algorithm for digital signal interpolation and a novel architecture to implement the algorithm in an integrated circuit (“IC”) with significant space constraints are presented. According to embodiments of the present invention, the interpolator is divided into two parts. The first part of the interpolator increases the sample rate by a factor of two and smoothes the signal using a half-band Infinite Impulse Response (“IIR”) filter. The second part of the interpolator increases the sample rate of the signal by a factor of thirty-two using a zero-order-hold (“ZOH”) circuit. In one embodiment, the half-band IIR filter is implemented using an all-pass lattice structure to minimize quantization effects. The lattice coefficients are chosen such that the structure can achieve all filter design requirements, yet is capable of being implemented with a small number of shifters and adders, and no multipliers.

Abstract: In a system having an DSP, an ASIC and a memory, in which the ASIC generates a number of different competing interrupts for the DSP to service, the ASIC has an interrupt request control module which automatically provides the DSP with a vector pointing to the memory location of the interrupt service routine for the currently pending interrupt request having the highest priority of all pending requests. The DSP reads this vector and uses it to access the interrupt service routine in the memory. Reading of this vector causes the interrupt request to be de-asserted, which causes the next highest priority pending interrupt request to become the highest priority pending interrupt request. As a result, a new vector is presented for the next read by the DSP.

Abstract: A simplified algorithm for digital signal interpolation and a novel architecture to implement the algorithm in an integrated circuit (“IC”) with significant space constraints are presented. According to embodiments of the present invention, the interpolator is divided into two parts. The first part of the interpolator increases the sample rate by a factor of two and smoothes the signal using a half-band Infinite Impulse Response (“IIR”) filter. The second part of the interpolator increases the sample rate of the signal by a factor of thirty-two using a zero-order-hold (“ZOH”) circuit. In one embodiment, the half-band IIR filter is implemented using an all-pass lattice structure to minimize quantization effects. The lattice coefficients are chosen such that the structure can achieve all filter design requirements, yet is capable of being implemented with a small number of shifters and adders, and no multipliers.