Abstract: Digital audio tone control implemented using Shelving filters for the digital audio treble tone control exhibits artifacts (noise, distortion, etc.) as the tone control settings are changed. This was previously accomplished by changing filter coefficients in the traditional small equal (on a dB scale) filter steps of a fraction of 1 dB. While this worked for bass filters, artifacts were still present for treble. This invention eliminates these artifacts by changing the filter steps to small equal steps on a linear scale. Additionally, where the steps became too large for the resolution required, additional filter steps are added. Approximately 150 filter steps are used for treble control and 128 filter steps are used for bass tone control. An efficient way of implementing the filter steps for digital tone control stores (119) one set of filter coefficient values and a small amount of additional information and then increments the coefficients between all the other steps.

Abstract: A method for generating digital filters for tuning a hearing aid to enhance hearing ability. Data is provided, for a range for a target response curve representative of enhanced hearing ability of sound level versus frequency. Second digital data is also generated, for an initial response curve or audiogram of an individual's hearing ability. The first digital data is compared with the second digital data and it is determined whether the initial response curve is within the tolerance range. If not, digital audio filters are iteratively generated, and the digital audio filters are applied to digital data representing received sound signals to generate third digital data to enhance hearing. Parameters of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range or a predetermined number of digital audio filters has been reached.

Abstract: An electronic dB-to-linear gain conversion system (10). The system comprises an input (12) for receiving a gain index signal (GI) representing a desired dB value. The desired dB value is selected from a set having an integer number S of dB values. The system also comprises a storage circuit (16) for storing an integer number V of linear gain values and circuitry for producing a linear gain signal (LG) in response to the gain index signal and to one of the V linear gain values. In the preferred embodiment, V is less than S.

Abstract: In a microprocessor, a method for providing a sample-rate conversion (“SRC”) filter on an input stream of sampled data provided at a first rate, to produce an output stream of data at a second rate different from the first rate. The input stream of sampled data is operated on with a first low-order interpolation filter routine to produce a first stream of intermediate data. The first stream of intermediate data is operated on with a first simplified interpolation filter routine, having a substantially small number of operations to calculate the coefficients thereof, to produce a second stream of intermediate data. The second stream of intermediate data is operated on with a first decimating filter routine to produce the output stream of data.

Abstract: A signal generator (10). The signal generator comprises circuitry (20) for producing at least a first input noise signal (N1), wherein the first input noise signal has a statistically insignificant autocorrelation.

Abstract: A digital audio dynamic range compressor includes a root mean square estimator receiving first and second audio input samples and generating root mean square values of the samples. A gain calculator receives the root mean square values and computes a gain for each input sample in the linear domain, not in the logarithmic or dB domain. A minimum selector receives the computed gain of each input sample and determines a minimum. An attack and release filter receives the minimum gain value and filters the minimum gain value according to attack and release coefficients and generate a gain output. A multiplier receives the gain output and multiplies the first and second audio input samples with the gain output.

Abstract: A system (10) for providing an integer number N of filters. The system comprises an input (Di) for receiving a digital audio signal and an output (Do) for providing a filtered audio signal. The system also comprises circuitry (16) for storing at least a first set of fixed filter coefficients and circuitry for storing estimation data. The system also comprises circuitry (14) for estimating a number of sets of estimated filter coefficients in response to the estimation data and the fixed filter coefficients. The system also comprises circuitry (14) for applying a transfer function to the digital audio signal and in response for providing the filtered audio signal. The circuitry for applying the transfer function applies a set of filter coefficients selected from the first set of fixed filter coefficients and the sets of estimated filter coefficients. Also, the transfer function is selected from a transfer function set consisting of a high pass filter transfer function and a low pass filter transfer function.

Abstract: A method for generating digital filters for equalizing a loudspeaker. First digital data is provided, for a tolerance range for a target response curve of sound level versus frequency for the loudspeaker. Second digital data is generated, for an actual response curve of sound level versus frequency for the loudspeaker. The first digital data is compared with the second digital data and it is determined whether the actual response curve is within the tolerance range. If the actual response curve is not within the tolerance range, digital audio filters are iteratively generated, and the digital audio filters are applied to the second digital data to generate third digital data for a compensated response curve. The frequency, amplitude and bandwidth of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range or a predetermined limit on the number of digital audio filters has been reached, whichever occurs first.

Abstract: A digital signal system (10, 100) for determining an approximate reciprocal of a value of x. The system includes an input (12) for receiving a signal, and circuitry (18) for measuring an attribute of the signal. The measured attribute relates at least in part to the value of x. The system further includes circuitry (104) for identifying a bounded region within which x falls. The bounded region is one of a plurality of bounded regions, and each bounded region has a corresponding slope value and first and second endpoints. The system further includes circuitry (106, 108, 110) for determining the approximate reciprocal by adjusting a reciprocal value at one of the first and second endpoints by a measure equal to a distance of the value of x from the one of the first and second endpoints times the slope value corresponding to the bounded region within which x is identified as falling.

Abstract: A digital graphic equalizer uses a predetermined number of equalizing bands each having a different center frequency, and the center frequencies span a predetermined audio bandwidth. For each equalizing band a minimum set of filters is provided. The filters have a predetermined linear uniform spacing between the gain of successive filters.

Abstract: A digital signal system (30) for determining an approximate logarithm of a value of x having a base b. The system comprises circuitry (32) for storing x as a digital representation and circuitry for identifying a most significant digit (MSD) of the digital representation. Adjacent the most significant digit is located a set of bits in respective lesser significant bit locations. The system further comprises a table (36) for storing a set of predetermined logarithms having the base b, wherein each of the predetermined logarithms corresponds to a number in a set of numbers. The system further comprises circuitry for addressing the table in response to a first bit group (t) of the set of bits in respective lesser significant bit locations, and in response the table is for outputting a one of the predetermined logarithms corresponding to a first number (Ia) in the set of numbers.

Abstract: A digital signal system (50) for determining an approximate antilog x from a value of ƒ(x), wherein x has a base b. The system comprises circuitry (52) for storing the value of ƒ(x) as a digital representation, wherein the value of ƒ(x) has an integer portion and a decimal portion (i.f). The system also comprises circuitry (53) for setting a most significant digit bit position MSD of the approximate antilog x in response to the integer portion of ƒ(x), wherein adjacent the most significant digit bit position MSD is located a set of bits in respective lesser significant bit locations. The system also comprises a table (36) for storing a set of predetermined logarithms having the base b, wherein each of the predetermined logarithms corresponds to a number in a set of numbers.

Abstract: A method of implementing a digital filter utilizes a fast and accurate reciprocal estimate function to generate and recharacterize the digital filter on the fly. The reciprocal estimate function operates to synthesize a digital filter rapidly and efficiently without the necessity for trigonometric and/or division capabilities, thereby preserving integrated circuit real-estate.

Abstract: In a microprocessor, a method for providing a sample-rate conversion (“SRC”) filter on an input stream of sampled data provided at a first rate, to produce an output stream of data at a second rate different from the first rate. The input stream of sampled data is operated on with a first low-order interpolation filter routine to produce a first stream of intermediate data. The first stream of intermediate data is operated on with a first simplified interpolation filter routine, having a substantially small number of operations to calculate the coefficients thereof, to produce a second stream of intermediate data. The second stream of intermediate data is operated on with a first decimating filter routine to produce the output stream of data.

Abstract: Systems and methods for determining coefficients of an FIR filter are disclosed. The FIR filter coefficients are computed by determining a sine of an input value and an inverse of the input value. The sine of the input signal and the inverse of the input signal are multiplied together to form a sinc value of the input value. The sinc value is employed to determine the coefficient. The system and method can be repeated to compute any number of FIR filter coefficients in real-time. The sine of the input signal is computed utilizing a memory lookup table. The memory lookup table includes pairs of uniformly distributed values for the sine and cosine functions in the range of 0 to &pgr;. The inverse of the input value is computed using an inverse memory lookup table, a most significant digit and a remainder. The coefficient is then computed from a product of the sine of the input signal and the inverse of the input signal.

Abstract: A simplified design for a family of bass, treble and graphic equalizer filters (20). The filter design allows on-the-fly implementation of these simplified filter types, even in audio systems without extensive computation resources. The methodology includes designing a nominal filter, and decomposing this nominal filter into a simplified transfer function such that the family of treble boost, cut, and equalizer filters can be realized with moderate computational resources.

Abstract: A method for providing a multi-stage filter on an input stream of digital data. In the method, the input stream of digital data is operated on with a first polyphase filter routine to produce a first stream of intermediate data. The first stream of intermediate data is operated on with a second polyphase filter routine. An optimizing indexing procedure is applied in performing instructions of the routines so as to execute fewer instructions that do not generate intermediate data on which the output stream of data is based.

Abstract: A method for generating digital filters for tuning a hearing aid to enhance hearing ability. Data is provided, for a range for a target response curve representative of enhanced hearing ability of sound level versus frequency. Second digital data is also generated, for an initial response curve or audiogram of an individual's hearing ability. The first digital data is compared with the second digital data and it is determined whether the initial response curve is within the tolerance range. If not, digital audio filters are iteratively generated, and the digital audio filters are applied to digital data representing received sound signals to generate third digital data to enhance hearing. Parameters of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range or a predetermined number of digital audio filters has been reached.

Abstract: A high performance audio processing device having significantly increased reconfigurability capabilities and that offers far more precision and throughput than its predecessors without increasing the per-channel cost of the device above the per-channel cost of the predecessors. The device architecture includes a single-cycle, 48×28 bit multiply feature that is configured as a sophisticated switch to minimize the use of branching such that the reconfiguration capabilities of the device are significantly increased. Multiplying by zero shuts a path off, while multiplying by one turns the path on, and multiplying by −1 turns the path on, but with an inversion of the path signal's phase. The use of a full-precision multiply rather than other forms of decision logic also allows “morphing” from one setting to another on-the-fly, and provides new operating states that are linear combinations of the “on” and “off” settings.

Abstract: A method for providing a sample-rate conversion (“SRC”) filter on an input stream of sampled data provided at a first rate, to produce an output stream of data at a second rate different from the first rate. The input stream of sampled data is operated on with a first low-order interpolation filter routine to produce a first stream of intermediate data. The first stream of intermediate data is operated on with a first simplified interpolation filter routine, having a substantially small number of operations to calculate the coefficients thereof, to produce a second stream of intermediate data. The second stream of intermediate data is operated on with a first decimating filter routine to produce the output stream of data.