Abstract: A sound reproduction system comprising a digital audio signal input (1), a digital audio signal processor (2, DSP) and a digital audio signal output (3) wherein the digital signal processor (2, DSP) comprises a high pass (HP) filter (21) with a high pass frequency (f), an amplifier (22) for a signal filtered by the HP filter, and a low pass (LP) filter (23) with a low pass frequency (f) for filtering the signal after amplification by the amplifier (22) and for providing an output signal, and the digital processor comprises an establisher (24, 25) for establishing the high pass frequency or the low pass frequency and a matcher (26) for matching the high pass frequency and low pass frequency of the high pass filter and low pass filter respectively to each other.

Abstract: A digital audio signal processing system and method transforms two-channel stereo time-domain data into the frequency domain. Vector operations are performed upon the frequency-domain data by which signal components unique to one of the input channels are routed to one of the output channels, signal components unique to the other of the input channels are routed to another of the output channels, and signal components common to both channels are routed to a third and optionally to a fourth output channel. The frequency-domain output channels are then transformed back into the time-domain, forming an equivalent number of channels of output audio data. The vector operations are performed in a manner that preserves the overall information content of the input data.

Abstract: Input audio signals are divided into low-frequency components and high-frequency components, both of which are subjected to delay processing in correspondence with desired positions of focal points with respect to speaker units respectively. Delayed low-frequency components are further subjected to weighting using a first window function. Delayed high-frequency components are subjected to weighting using a second window function (e.g., a Hamming window function). Weighted high-frequency components and weighted low-frequency components are added together with respect to the speaker units, which are thus driven respectively. The first window function applied to low-frequency components is made moderate in weighting in comparison with the second window function applied to high-frequency components; thus, it is possible to reduce differences of sound directivities between low-frequency components and high-frequency components of audio signals.

Abstract: An amplifier is provided using digital potentiometer integrated circuits to control the tone of a vacuum tube preamplifier allowing digital control of the analog signal path of the amplifier. Using digital potentiometer integrated circuits to control the tone of a vacuum tube preamplifier results in an amplifier that preserves the unique tone quality of a vacuum tube amplifier that offers the flexibility, versatility, and user-friendly features of a digitally controlled amplifier, such as the ability to save and recall amplifier settings. The amplifier of the present invention is especially applicable for use with musical instruments such as for example, electric guitars.

Abstract: An integrating apparatus to lower the levels of audio outputs of an on-board audio system immediately upon cessation of a noise level such as when a vehicle comes to a stop. Audio signals from an audio source are amplified through attenuating means by an amplifying means to drive a loudspeaker. An output of a microphone for detecting a noise is inputted to each of two integrating circuits, which have different fall time constants. Outputs of the integrating circuits are applied to a selectively outputting means, which selectively derives an output having a lower level among the outputs of the integrating circuits, and which then supplies the output as a control signal to the attenuating means. When the level of noise is high such as when the vehicle is running, the control signal has a high level and the attenuating means reduces an attenuation. Thus, the output of the audio source may be heard without a drift of the audio signal level.

Abstract: An audio signal processing device (1) is arranged for providing a level controlled output signal (So) in response to a selected input signal (Si). The device comprises a selection unit (11) for selecting an input signal (Si) from a plurality of input signals, an amplification unit (12) for amplifying the selected input signal so as to produce the output signal (So), and a gain control unit (15) for controlling the gain of the amplification unit (12) in response to a level signal (Sl) indicative of the audio signal level. The device is arranged for deriving the level signal (Sl) from the output signal (So) instead of the input signal (Si) of the amplifier (12). As a result, any signal distortions introduced by the amplifier (12) have no effect on the gain control and a greater precision is achieved.

Abstract: A system is provided for receiving audio signals from a plurality of microphones and transferring the audio signals via a common composite signal channel to a receiving unit, such as a mixing console. The system has at least two satellite units, each having a microphone signal input, a composite signal channel input and summing means for summing a microphone signal and a composite signal. The system further has a master unit with a composite signal channel input, and signal converting means for converting said composite signal into a master signal, which is provided to the receiving unit via a master signal output. In the system, each satellite unit is connected to the common composite signal channel, such that the microphone signal received at the respective satellite unit, is added to the composite signal, which is fed to the master unit.

Abstract: Systems, devices, and methods are provided to inhibit apparent amplitude modulation in non-linear processing that causes distortion in a processed signal. One aspect of the invention includes a hearing aid. The hearing aid includes a microphone to receive an input signal, a speaker to reproduce the input signal, and a processor. The processor processes the input signal using a gain. The processor includes an inhibitor, which inhibits distortions, and an adjuster, which adjusts the gain. The inhibitor acts to smooth an envelope of the input signal to inhibit undesired modulation. The adjuster adjusts the gain if the envelope is either above or below a threshold.

Abstract: Microphone equalization is implemented in a directional microphone system of the second or a higher order with at least three omnidirectional microphones. Initially, the signal levels of the microphone signals generated by the three omnidirectional microphones are compensated. The phase is subsequently varied in one of the three microphone signals until the signal levels of the directional microphones of the first order that are formed from the three omnidirectional microphones, are also compensated.

Abstract: The present invention is related to a method for adjusting the frequency characteristic of a digital audio processor having adjustable parameters including at least low frequency gain, high frequency gain and volume gain. The invention is also related to an audio device executing the inventive method every time the characteristic frequency response curve is adjusted. Today's audio devices frequently offer the feature that the user can select among a group of different sound characteristics. Typically the user may want to adapt the sound characteristic of the device to the kind of music he is listening, e.g. rock, jazz, or classic music. In technical terms the sound characteristic is widely determined by the characteristic frequency response curve of the audio signal processing. Those characteristic curves are essentially defined by a gain value in the low, middle and high frequency range. Normally, a 3-band audio processor, which allows adjusting these three parameters independently is used for this purpose.

Abstract: An electronic volume device of an invention includes a common volume I/O 371 for receiving an i-th common volume level Dcom[i] (i=1 to N) on a common scale of N steps, a volume converter 373 for converting the Dcom[i] into a j-th unique volume level Dvol[j] on a unique scale of n steps; and a volume controller 372 for controlling an electronic volume based on the unique volume. The volume converter 373 includes a volume memory for storing an offset between the Dcom[i] and the Dvol [j] and a volume controller that controls at least one of the Dvol[j] and the offset such that the received Dcom agrees with the sum of the Dvol[j] and the offset and controls the Dvol[j] on a higher priority than the offset.

Abstract: A method and apparatus for generating and controlling volume of a speaker of an appliance is disclosed. The appliance includes an IC chip connected to an amplifier subsystem. The IC chip includes a square-wave audio signal generator, a counter, a register, a comparator, and an AND gate. Theses components of the IC chip are used to generate modulated audio frequency square-wave signal. The modulated audio frequency square-wave signal having pulses, each pulse has a width determined by the volume control value. The modulated audio frequency square-wave signal is sent from the IC chip to the amplifier subsystem on a single connection. At the amplifier subsystem, the modulated audio frequency square-wave signal is integrated over, filtered, and amplified to drive a speaker to produce the desired sound. By adjusting the volume control value, the widths of the pulses, thus the volume of the produced sound can be controlled.

Abstract: In a frequency characteristic adjustment apparatus incorporated in an audio system, a frequency characteristic of an audio signal is made to agree with a target frequency characteristic, thus providing an equalizing function. In the apparatus, by a frequency characteristic controller, a measurement signal is detected as a sound-collected signal that received by way of speakers/microphone, and the sound-collected signal is divided into a signal component falling into a one fixed-level band and one or more signal components falling into one or more variable-level bands. A relative level of the signal component in each variable-level band is estimated on the basis of a level of the signal component in the fixed-level band in the target frequency characteristic. Based on the estimated relative levels, the levels at equalizers are set, so that the levels of signal components of the signal are adjusted by the equalizers channel by channel.

Abstract: To improve upon a method of adjusting at least one filter parameter such as the mid-frequency of at least one filter unit provided with a playback system for audio signals, this filter unit being provided in particular for at least one digital equalizer assigned to the playback system, as well as an associated playback system so as to achieve automatic optimization of an acoustically correct bass level boosting, [a] first the acoustic frequency response is determined, [b] then a cutoff frequency below which the sound pressure falls below a sound pressure threshold level based on the average sound pressure is determined, and [c] next the filter parameter(s) of the filter unit is/are adjusted automatically, so that the mid-frequency of the filter unit is just above the cutoff frequency thus determined.

Abstract: Arrangements are made such that a plurality of operators, such as faders, can be set to opposite operational directions. When any one of the operators has been operated in a particular direction, a control value of each of the other operator, set to the same operational direction as the operated operator, is controlled in the same direction as the particular operated direction, while a control value of each of the other operators, set to the opposite operational direction from the operated operator, is controlled in the opposite direction from the particular operated direction. Where control values of a plurality of channels are to be controlled via a collective control controller, each of the channels is set to a forward or inverse operational direction. Thus, even faders that are to be operated in opposite directions can be appropriately operated collectively as a group.

Abstract: A method for processing audio signal that includes receiving a plurality of digital audio signals and calculating signal levels of them. The next step is calculating smooth attenuation signal for each one of the input audio signals. The smooth attenuation signals are calculated according to the signal levels and smoothing criteria, such that the mixing of signals that include the input audio signals multiplied by their respective smooth attenuation signals is peak limited by a given threshold level.

Abstract: An audio signal processing apparatus includes a high-pass filter for extracting from an audio signal a frequency component higher than f0, n band-pass filters for extracting, from the audio signal, frequency components falling within a frequency range from f0/N to f1/N, n harmonic overtone generators for frequency multiplying each of outputs of the n band-pass filters by N, a first combining unit for combining the generated harmonic overtone components, a level detector for detecting a level of a supplied harmonic overtone component, a gain controller for controlling dynamically the harmonic overtone component supplied from the first combining unit, and a second combining unit for combining the frequency component extracted by the high-pass filter and a harmonic overtone component output from the gain controller, where N is (n being a natural number), f0 is a first predetermined frequency, and f1 is a second predetermined frequency higher than f0.

Abstract: This invention relates to a sound system and method for processing acoustical sound. The sound system comprises a microphone for converting an acoustical sound to an electrical sound signal, a processor for processing the sound signal and for generating a processed sound signal, and a speaker for converting the processed sound signal to a processed acoustical sound.

Abstract: A method for influencing, e.g., for decreasing, the treble reproduction of an audio signal for reproduction obtained from a received signal, the received field strength and reception quality of the received signal being evaluated and, as a function thereof, the transfer function of at least one filter unit that can be impinged upon by the audio signal being controlled, as well as a circuit assemblage or arrangement for performing such a method for the modification of treble reproduction, so that, among other things, reproduction of the interference may always be reliably concealed and so that no noticeable and/or irritating “flutter” effects may occur. The method may include processing of the output signals of at least two reception interference detectors, and processing of the received field strength may be accomplished respectively in at least one first processing branch operating at a first, e.g.

Abstract: A normalization for streaming digital audio signals applies a gain factor according to the maximum sample magnitude in a window of samples and compare the gain factor to prior gain factors to adjust the gain factor for the samples in the window of samples. Adaptation of the gain factor with rapid decreases but slow increases avoids saturation but allows quiet passages.

Abstract: A system and method for automatically controlling gain in a signal chain are presented. A first signal is processed to produce a processed signal. A gain is adjustably applied to the processed signal using an automatic gain control (AGC) mechanism. A resulting signal is produced. The gain is adjusted based on the first signal. As such, even when a desired signal is not substantially present in the first signal, gain is not excessively applied to the processed signal.

Abstract: A volume control apparatus and a method enable the volume of sound output from a sound output system to be adjusted in units of an increase/decrease interval set by a user, the volume control apparatus controlling the volume of sound output from the sound output system on a basis of volume level information set by the user.

Abstract: An electronic equipment is provided that includes an audio reproduction circuit for receiving an audio signal and reproducing the audio signal in accordance with a preselected audio frequency gain profile. In addition, the electronic equipment includes a volume gain control circuit for providing user adjustable volume gain to the audio signal reproduced by the audio reproduction circuit. The electronic equipment also includes an audio output for outputting the reproduced audio signal. The audio reproduction circuit alters the preselected audio frequency gain profile as a function of the user adjusted volume gain.

Abstract: Some embodiments of the invention provide a method for controlling the volume of an audio track. This method represents the volume of an audio track with a graph. This graph is defined along two axes, with one axis representing time and the other representing the volume level. A user can adjust the graph at different instances in time in order to change the volume level in the audio track at these instances. Different embodiments use different types of graphs to represent volume. For instance, some embodiments use a deformable line bar.

Abstract: The present invention provides a sound reproduction device and method for automatic gain control. The method includes steps of: providing a gain index table for listing a plurality of time period each referencing a predetermined gain value; providing a plug for attaching to an audio signal source, receiving audio signals from the audio signal source, and generating a request signal; receiving the request signal generated by the plug; reading a current time and determining a time period corresponding to the current time; reading a predetermined gain value referenced by the time period in the gain index table; replacing the default gain value with the predetermined gain value referenced by the time period; and amplifying the audio signals received from the audio signal source by the predetermined gain value referenced by the time period, thereby audio energy of the amplified audio signals is proper to a user's hearing.

Abstract: An audio signal processor includes a tone control (23). The tone control comprises two low-pass filters (221, 222) operating in current-mode and a subtractor (223) which subtracts the output currents of the filters to produce a band-pass characteristic. Each filter is a tuneable log-domain current-mode filter comprising MOS transistors operating in weak inversion. The tone control is useful in audio signal processors, hearing aids and single-channel and multi-channel Cochlear implants.

Abstract: A method for use with equipment having filters, the method characterized by multiplying a matrix having a specified cut-boost setting of a filter of a graphic equalizer by a correction matrix to create a matrix having a corrected cut-boost setting of the filter of the graphic equalizer; adjusting an actual cut-boost setting of the filter of the graphic equalizer to be substantially equal to the corrected cut-boost setting of the filter of the graphic equalizer; and configuring the filter to have a Q value substantially equal to a linearizing Q value. In addition to the foregoing, other method embodiments are described in the claims, drawings, and text forming a part of the present application.

Abstract: A system for enhancing sound quality comprising a filter that square roots the instantaneous amplitude of frequencies in an input signal for generating artificial harmonics corresponding to said frequencies. The system can comprise an automatic level control that momentarily boosts the amplitude of artificially generated harmonics to emphasize attack transients occuring within the input signal.

Abstract: An audio input signal is processed to produce a processed audio output signal. An audio input signal is received as an original signal. The audio input signal is dynamically filtered to produce a first stage signal consisting of a selected frequency band of the input signal. Gain applied to the first stage signal is dynamically controlled in response to a control signal to produce a second stage signal. The control signal is derived from the first stage signal. Processing the original signal in combination with the second stage signal to produce a processed audio output signal. Processing the original signal in combination with the second stage signal and the first stage signal to produce a processed audio output signal.

Abstract: Audio processing methods and apparatus are provided for at least partially compensating for the Fletcher-Munson effect. An audio processor includes a variable filter receiving an input signal and providing a filtered output signal, the variable filter having a fixed cutoff frequency and a quality factor that is controllable in response to a control signal, and a control circuit configured to detect a signal level representative of input signal level in a selected band and to generate the control signal in response to the detected signal level. The control circuit may include a low-pass band select filter and a detector for detecting a signal level in the band selected by the low-pass filter and for generating the control signal.

Abstract: The invention is directed to a sound system with four loudspeakers and at least one signal processing device connected to the input of two loudspeakers and controlled by two input signals. The signal processing device produces control signals for the loudspeakers. The respective control signals are equal to the sum of the input signals which are weighted differently with respect to amplitude and phase, so that the radiation characteristic of the two loudspeakers depends on the weighting of the input signals.

Abstract: Conventional analog loudspeakers have a limited dynamic range as compared to the available dynamic range of digital recordings. Digital recordings use up to 24 bits and this implies a dynamic range of 141 dB. Digital loudspeakers, involving 2N single bit devices (with N=24, this number is 1.7×107) have been proposed. The present improvement is the provision of at least one loudspeaker, a plurality of analog drivers and the audio input supplied to a control processor which in turn drives one or more of the plurality of independent analog drivers. The number of drivers in operation at any one time is determined by the amplitude of the input audio signal to the control processor.

Abstract: A sound signal reproducing apparatus for supplying sound signals of a plurality of channels to speakers via sound amplifier circuits includes: a power supply voltage controller for controlling a power supply voltage to the sound amplifier circuits; a plurality of filter circuits for passing predetermined frequencies of the output signal of each of the sound amplifier circuits in the plurality of channels; a plurality of level detection circuits for detecting levels of output signals of the plurality of filter circuits; an adder circuit for adding output signals of the plurality of level detection circuits; and a comparator circuit for comparing an output signal of the adder circuit with a reference level, so that an output signal of the comparator circuit controls the power supply voltage controller.

Abstract: The present invention relates to a process for adjusting the sound volume of a digital sound recording characterized in that it comprises: a step consisting of determining, in absolute values, for a recording, the maximum amplitude values for sound frequencies audible for the human ear, a step consisting of calculating the possible gain for a specified sound level setting, between the maximum amplitude value determined above and the maximum amplitude value for all frequencies combined, a step consisting of reproducing the recording with a sound card by automatically adjusting the amplification gain level making it possible to obtain a sound level for the recording of a specified value so that it corresponds to the gain calculated for this recording.

Abstract: A method and apparatus for applying a gain characteristic to an audio signal are provided. Data storing a plurality of gain characteristics at a plurality of different levels is stored in a storage means. The amplitude of an input signal is repeatedly assessed and from this a gain characteristic to be applied to the input is determined.

Abstract: In correcting the sound field, the loudspeakers 6FL to 6WF are sounded by the noise. The attenuation factors of the inter-band attenuators ATF11 to ATFki for adjusting gains of the band-pass filters BPF11 to BPFki to the frequency in respective channels are corrected based on detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factors of the channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the delay times of the delay circuits DLY1 to DLY5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factor of the channel-to-channel attenuator ATGk is corrected based on the detection result of the reproduced sound of the loudspeaker 6WF as the subwoofer, whereby the levels of the reproduced sounds reproduced by the loudspeakers 6FL to 6WF are adjusted to be made flat over the audio frequency band.

Abstract: A filter coefficient of a graphic equalizer GEQ is corrected based on detection results of reproduced sounds generated by supplying a noise to all frequency band loudspeakers 6FL to 6RR and a low frequency band exclusively reproducing loudspeaker 6WF via the graphic equalizer GEQ. Then, attenuation factors of channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6FL to 6RR via the graphic equalizer. Then, delay times of delay circuits DLY1 to DLYk are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6FL to 6WF via the graphic equalizer.

Abstract: Maximal and minimal values represented by samples of a digital audio signal are detected. A number of samples from a sample representing a minimal value to a maximal-value-corresponding sample is detected. A number of samples from a sample representing a maximal value to a minimal-value-corresponding sample is detected. Calculation is given of a first difference between the maximal-value-corresponding sample and the immediately-preceding sample. Calculation is given of a second difference between the minimal-value-corresponding sample and the immediately-preceding sample. First and second coefficients are calculated from the detected sample numbers. The first coefficient and the first difference are multiplied to generate a first multiplication result. The second coefficient and the second difference are multiplied to generate a second multiplication result. The maximal value represented by the maximal-value-corresponding sample is incremented by the first multiplication result.

Abstract: A method of enhancing the audio quality in a reproduction medium having unknown characteristics. With this method a predetermined finite set of single frequency tones are generated and these tones are then passed through the reproduction medium to generate an output signal, which in turn is passed through a set of sub-band filters. Each of the sub-band filters pass at least a frequency corresponding to one of the tones in the set of tones. The characteristics of the reproduction medium is then estimated as a result of passing the output signal through the set of sub-band filters. Based on the estimated characteristics of the reproduction medium, a set of sub-band inverse filters are constructed. Finally before passing the audio signal through the reproduction medium the signal is passed through the set of inverse filters to improve the quality of the audio signal after it passes through the reproduction medium.

Abstract: An earmuff comprising a headpiece and a circuit. The headpiece supports the earmuff on the head of an individual. The circuit has an input device for receiving external sound energy and converting the external sound energy to electrical sound signals.

Abstract: In correcting the sound field, loudspeakers 6FL to 6WF are sounded by the noise. The attenuation factors of the inter-band attenuators ATF11 to ATFki for adjusting gains of the band-pass filters BPF11 to BPFki to the frequency in respective channels are corrected based on detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The attenuation factors of channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The delay times of delay circuits DLY1 to DLY5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The attenuation factor of a channel-to-channel attenuator ATGk is corrected based on the detection result of the reproduced sound of the loudspeaker 6WF as the subwoofer. Therefore, the levels of the reproduced sounds reproduced by the loudspeakers 6FL to 6WF are adjusted to be made flat over the audio frequency band.

Abstract: Minimizing the damaging effects of loud music but with minimal reduction in aesthetic quality of the music. When music presented through an audio is sampled in order to determine average frequency, music having very low average frequencies is predictably likely to be played louder than sampled music having higher average frequencies. This unexpected observation is based upon a human factor, i.e. rock and related music that is ordinarily composed and presented at relatively low frequencies, e.g. 500 to 1000 Hz is likely to be played on the audio system at very high volume because of the tastes of its listeners. An audio system for a data processor controlled presentation of music with a set up for minimizing the damaging effects of loud music comprising an implementation for predetermining a minimum average frequency of music being presented in combination with measuring whether the music being presented falls below said minimum average frequency for a sampling period.

Abstract: The invention is a method, apparatus and computer readable program for eliminating overflow and underflow by re-adjusting or rectifying the audio levels of an audio signal in those portions where overflow/underflow occurs. In a first preferred embodiment a single stage scale down solution is provided in starting and end points of the overflow/underflow areas in the audio signal are detected, and a number of discrete sampling points along the audio signal curve between these two points are adjusted by a predetermined multiplier in order to reduce the positive level (in the case of overflow) or negative level (in the case of underflow) of each discrete sampling point. Using the single stage scale down solution of the present invention, a new audio signal is created having no overflow/underflow, yet maintaining the former generally smooth shape of the audio signal waveform and signal characteristics, without completely clipping the audio levels.

Abstract: A system and method providing content for access by a portable device at an event is provided. The method includes receiving a plurality of signals from a plurality of sources and combining at least some of the plurality of signals for transmission to a plurality of broadcast systems, with the plurality of broadcast systems corresponding to a plurality of events. The method further includes determining the broadcast systems to which each of the combined signals is to be transmitted, transmitting the combined signals to at least one of the determined broadcast systems, and broadcasting the combined signals received by at least one of the broadcast systems corresponding to at least one event. The combined signals are configured to allow selection of content with a portable device, with the content defined by at least one of audio and video signals of the combined signals.

Abstract: Acoustical characteristics of rooms or other environments in which audio systems operate may be improved by equalization derived from measures of relative acoustic energy storage of the audio system such as a Relative Acoustic-energy Decay Spectrum (RADS). A RADS may be calculated from a measure of absolute energy storage such as RT60 values obtained from models or measurements of the audio system, and normalized with respect to a measure of acoustic energy storage of a reference system.

Abstract: An improved hearing aid is disclosed having a switched release automatic gain control. In one embodiment, an input signal is presented to a comparator with a reference typically around, for example, 65 dB SPL instantaneous. The output of the comparator controls a switch in a release circuit of an automatic gain control. While the instantaneous input level exceeds the threshold, the release circuit is enabled and the automatic gain control behaves normally. While the instantaneous input level does not exceed the threshold, the release circuit is disabled and the automatic gain control maintains its current gain setting, essentially indefinitely. Because speech, even soft speech, has a very high positive peak content, this circuit will recover any needed gain in the presence of speech (assuming an appropriate threshold) but not recover gain to background noise whose positive peak content is below the threshold.

Abstract: A circuit for providing a variable amount of bass control on an input signal dependent on a signal level of the input signal includes a high-pass filter, a variable impedance shunt connected to the output of the high-pass filter, and a signal level detector for controlling the variable impedance shunt. Depending on the signal level, the variable impedance shunt varies the Q value of the high-pass filter so that at low signal levels, the Q value is at its maximum while at high signal values, the Q value is lowered.

Abstract: A method for controlling audio gain balance in a multi-mode communications device (200) includes providing (201) microphone audio input to the multimode communications device. The audio is supplied to an amplifier gain stage while a dynamic instantaneous energy value of the audio input is computed (203) when in a first operational mode such as an analog mode. A predetermined gain algorithm is processed (207) representing an audio input when in a second mode such as a digital mode using the energy value. The audio gain stage in the multi-mode communications device is then adjusted (209) so that audio gain when in the first mode substantially approximates an audio gain when in an second mode. This provides a consistent audio amplitude output when receiving the first and second modes in a radio receiver.

Abstract: Difference compensation circuits are connected to a volume controller having a tap in series. Each of the difference compensation circuit comprises a voltage control amplifier and an external CR network. The difference compensation circuits are arranged so as to independently attenuate a bass range and increase a mid-range in accordance with an amount of volume control by the volume controller.

Abstract: Systems, devices, and methods are provided to inhibit apparent amplitude modulation in non-linear processing that causes distortion in a processed signal. One aspect of the invention includes a hearing aid. The hearing aid includes a microphone to receive an input signal, a speaker to reproduce the input signal, and a processor. The processor processes the input signal using a gain. The processor includes an inhibitor, which inhibits distortions, and an adjuster, which adjusts the gain. The inhibitor acts to smooth an envelope of the input signal to inhibit undesired modulation. The adjuster adjusts the gain if the envelope is either above or below a threshold.