Abstract: The present invention provides systems and methods to detect when hard decisions change for bit nodes of one or more layers of a layered LDPC decoder and to update accumulated partial syndrome calculations for those layers. As hard decisions of bit nodes are generated, they are compared with their previous values. If the hard decisions change, partial syndrome calculations are accumulated and updated for the layers having non-zero elements in one or more columns of the parity check matrix corresponding to the bit nodes of the changed hard decisions. If the hard decisions for the bit nodes are unchanged, the partial syndrome calculations for the corresponding layers are not updated. Changes to hard decisions of codewords are tracked and partial syndromes are flipped for the layers of the columns of the parity check matrix corresponding to the bit nodes of the changed hard decisions.

Abstract: The present invention provides systems and methods to detect when hard decisions change for bit nodes of one or more layers of a layered LDPC decoder and to update accumulated partial syndrome calculations for those layers. As hard decisions of bit nodes are generated, they are compared with their previous values. If the hard decisions change, partial syndrome calculations are accumulated and updated for the layers having non-zero elements in one or more columns of the parity check matrix corresponding to the bit nodes of the changed hard decisions. If the hard decisions for the bit nodes are unchanged, the partial syndrome calculations for the corresponding layers are not updated. Changes to hard decisions of codewords are tracked and partial syndromes are flipped for the layers of the columns of the parity check matrix corresponding to the bit nodes of the changed hard decisions.

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 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 (“SNR”) 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.

Abstract: 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.

Abstract: 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.

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 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.

Abstract: 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.