Sonic Innovations, Inc.

Other

Xiaoling Fang

Keith L Davis

Douglas M Chabries

Gerald A Wilson

Brad Giles

Darrell Rose

Martin R Johnson

Renyuan Huang

Michael J Nilsson

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 A(Z) and B(Z) along with a fixed delay, instead of a single filter W(Z) and a delay to represent the feedback path in each subband. The first filter, A(Z), is called the training filter, and models the static portion of the feedback path in isubband, 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, B(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 isubband caused by jaw movement or objects close to the ears of the user.

Subband acoustic feedback cancellation in hearing aids

A multi-band spectral subtraction scheme is proposed, comprising a multi-band filter architecture, noise and signal power detection, and gain function for noise reduction. In one embodiment, the gain function for noise reduction consists of a gain scale function and a maximum attenuation function providing a predetermined amount of gain as a function of signal to noise ratio (&ldquo;SNR&rdquo;) and noise. In one embodiment, the gain scale function is a three-segment piecewise linear function, and the three piecewise linear sections of the gain scale function include a first section providing maximum expansion up to a first knee point for maximum noise reduction, a second section providing less expansion up to a second knee point for less noise reduction, and a third section providing minimum or no expansion for input signals with high SNR to minimize distortion. According to embodiments of the present invention, the maximum attenuation function can either be a constant or equal to the estimated noise envelope. The disclosed noise reduction techniques can be applied to a variety of speech communication systems, such as hearing aids, public address systems, teleconference systems, voice control systems, or speaker phones. When used in hearing aid applications, the noise reduction gain function according to aspects of the present invention is combined with the hearing loss compensation gain function inherent to hearing aid processing.

Noise reduction apparatus and method

A method and an apparatus for minimizing latency in digital signal processing paths. One example is an active noise cancellation device. The system includes a digital closed feedback loop having a forward path and a feedback path. The forward path includes a compensation filter, a digital-to-analog converter, and an output transducer. The feedback path includes an input transducer, a feedback delta-sigma modulator, and a feedback sampling-rate converter. An input signal is processed in one of several ways into a processed digital input signal having a preselected intermediate sampling rate. Through the feedback path, an analog output signal is processed into a digital feedback signal having substantially the same preselected intermediate sampling rate. The processed digital input signal and the digital feedback signal are combined and processed through the forward path to produce an anti disturbance signal that is combined with a disturbance signal to form the analog output signal.

Method and apparatus for minimizing latency in digital signal processing systems

A method for fitting a hearing compensation device comprises selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness. The loudness levels may then be adjusted as needed to achieve perceived sameness across the frequency spectrum. A gain curve for each frequency is calculated from the selected plurality of loudness levels.

Hearing aid fitting system

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 A (Z) and B (Z) along with a fixed delay, instead of a single filter W (Z) and a delay to represent the feedback path in each subband. The first filter, A (Z), is called the training filter, and models the static portion of the feedback path in i 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, B (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 i subband caused by jaw movement or objects close to the ears of the user.

A decimation filter includes a zero insertion filter having an input and an output, wherein the input forms an input of the decimation filter; a cascade of four comb filters having a first accumulator stage and a second stage having a first differentiator portion and second differentiator portion wherein the first stage has an input coupled to the output of the zero insertion filter and an output, the first differentiator portion has an input coupled to the output of the first stage and an output, and the second differentiator portion has an input coupled to the output of the first differentiator portion and an output, and a frequency shaping filter having an input and an output, wherein the input is coupled to the output of the second differentiator portion and the output forms an output of the decimation filter.

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Draper, UT, United States of America

Xiaoling Fang