Abstract: A method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.

Abstract: A method for increasing ring tone volume is provided. The method includes steps of: reading an audio file which is set as a current ring tone; determining whether the ring tone is a MP3 audio file or a musical instrument digital interface (MIDI) audio file; adjusting frequencies by using an equalizer technique to increase volume of the ring tone if the ring tone is the MP3 audio file; adjusting a volume level of the ring tone to be the highest volume level, and adjusting timbre of the ring tone to increase the ring tone volume by simulating a musical score of the ring tone by using different instruments if the ring tone is the MIDI audio file. A related system is also provided.

Abstract: An audio signal processing apparatus and method for extracting components from an input signal, generating additional components, combining components, and level-controlling components. An audio signal processing apparatus may include a harmonic overtone adder and an equalizer. A harmonic overtone adder may include a high-pass filter, a low-pass filter, an harmonic overtone generator, and a combining unit. An equalizer may include a level detector and a gain controller.

Abstract: A system and method for dynamic bass frequency control, the system comprising: a bandwidth filter coupled to a plurality of switchable electronic components for filtering a frequency range of an input signal; an amplifier coupled to the bandwidth filter and the plurality of switchable electronic components, the amplifier being configured for adjusting amplitude of the filtered frequency range of the input signal based on a configuration of the plurality of switchable electronic components; and a signal level detector coupled to the switchable electronic components for determining signal strength of the input signal, the signal level detector being configured to switch the plurality of switchable electronic components between more than one configuration based on the signal strength of the input signal, each configuration determining the frequency range of the input signal to be filtered by the bandwidth filter and the amplitude adjustment to be made to the filtered frequency range of the input signal by the am

Abstract: Bass frequencies of audio can be boosted using various techniques and tools. The described techniques and tools can be applied separately or in combination. For example, bass frequencies of audio can be boosted using an integer bass boost filter by receiving user-settable parameters, such as “c” and “s” coefficients, and implementing the integer bass boost filter using a coupled form structure implementation and the user-settable parameters. Bass frequencies of audio can also be boosted using an integer bass boost filter that is configured to use any of plural coupled form structure implementations. Bass frequencies of audio can be also be boosted using a linear combination of an input audio signal and output of a high-pass filter.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: For hearing apparatus and especially for hearing devices a tone balance unit is to be able to be used even under difficult general conditions. Therefore a hearing apparatus with a signal processing device, through which signals are able to be processed in a number of frequency channels, and a control device for controlling the level of the individual frequency channels are provided. The setting device features an unsymmetrical tone balance unit related to a predetermined frequency with which at least one lower and at least one upper channel of the number of frequency channels related to a predetermined frequency is able to be simultaneously unsymmetrically adjusted relative to the predetermined frequency. Thus for example a tone balance unit can also be used with a low feedback limit in the upper frequency range.

Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a headliner speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: There is provided an automatic gain control circuit for volume control receiving at least one audio signal input via either a wired connection or a wireless connection. The circuit includes an output signal regulator receiving a first portion of the at least one audio signal input, with the regulator being able to vary an amplitude of the first portion of the at least one audio signal input when a control signal from a circuit controller is received. The circuit controller may be powered by a DC generator; with the DC generator receiving a second portion of the at least one audio signal input. A signal from the output signal regulator may be passed through an acoustic generator. It is preferable that the first portion of the at least one audio signal input is substantially more than the second portion. It is also preferable that the output signal regulator varies the amplitude of the at least one audio signal input by reducing the amplitude. A corresponding method of volume control is also disclosed.

Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.

Abstract: A device and method for improving the performance of an electro-acoustic transducer. An acoustic test signal is generated through the electro-acoustic transducer. The acoustic test signal is measured at multiple points on an ambient surface around the electro-acoustic transducer to create measured acoustic data. Based on the measured acoustic data, an acoustic power frequency response of the electro-acoustic transducer is calculated. A correction impulse response for the electro-acoustic transducer is determined based on the acoustic power frequency response. A correction filter applies the correction impulse response to a sound signal input to generate a sound signal output for playback through the electro-acoustic transducer.

Abstract: A speaker that includes a pair of spaced-apart, in-plane mounted drivers connected in series to a network for applying the appropriate frequency component of the electrical input drive signal to each of the drivers is disclosed. A frequency dependent shunting network, such as a low pass filter, is applied to one of the drivers so as to gradually mute the one driver as a desired frequency is approached. The result is an aesthetically pleasing speaker that has dual in-plane drivers and that produces superior sound quality throughout the entire frequency range of the speaker, including those ranges in which dual, in plane, drivers tend to acoustically interfere with each other.

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: A signal processing apparatus includes first and second extracting units extracting frequency components having a first frequency band and a second frequency band, respectively, from an input audio signal, a first-harmonic-component generating unit generating a first-harmonic-component signal including a frequency component whose frequency is N1 times that of the frequency component extracted in the first extracting unit, a second-harmonic-component generating unit generating a second-harmonic-component signal including a frequency component whose frequency is N2 times that of the frequency component extracted in the second extracting unit, and a combining unit combining the input audio signal, and the first- and the second-harmonic-component signals in a predetermined ratio. The first frequency band is higher than the second frequency band. N1 and N2 are positive integers, and N1 is smaller than N2.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: Methods, digital systems, and computer readable media are provided for determining a gain reduction parameter level for loudspeaker equalization by determining a noise score, an equalization effectiveness score, and an equalization non-effectiveness score for a candidate gain reduction parameter level, determining a composite quality score using the three scores, and designing a compensating filter for the loudspeaker using the candidate gain reduction parameter level if the composite quality score is better than composite quality scores of all other candidate gain reduction parameter levels.

Abstract: A variable equalizer apparatus for forward and/or reverse equalizers in an amplifier. The system can include a structure to allow continuous contact of the signal flow as an equalizer is removed; jumpers with fixed resistors and associated capacitors and inductors to produce a variable range over many different values; and/or variable resistance potentiometers with fixed resistors and associated capacitors and inductors to produce a variable range over separate value equalizers.

Abstract: An apparatus is provided for controlling a graphic equalizer which is implemented by digital filters, each being assignable to a frequency band for regulating a level of the frequency band. In the apparatus, a set of control devices are provided in correspondence to respective frequency bands, each control device being operable to specify the level of the corresponding frequency band to either of a reference level or other level than the reference level. An assignment section operates when a number of the digital filters is less than a total number of the frequency bands, for assigning the digital filters to the frequency bands which are specified with the levels other than the reference level. A control section controls remaining digital filters which are not assigned to any of the frequency bands in a through state.

Abstract: A system for boosting the bass of an audio signal to closely match or mirrors a plurality of Robinson-Dadson loudness curves by interpolating coefficients from a table of values representing the Robinson-Dadson loudness. The system having a controller that interpolates the coefficients from the loudness curves and then uses the coefficients in a shelf filter that makes adjustments to the audio signal. The result of the adjustments to the audio signal is the introduction of bass boost slowly through a diminuendo or lowering of level through volume adjustment and to removes the bass boost rapidly during a crescendo or increase in level through user volume adjustment.

Abstract: A method (200) for improving quality of output audio (126). The method can include detecting an output acoustic signal (128) and generating a receive audio signal (134) based, at least in part, on the detected output acoustic signal. A frequency domain representation (140) of the receive audio signal can be compared to a frequency domain representation (138) of a source audio signal (124) from which the output acoustic signal is generated. At least one distortion signal (142) in the receive audio signal can be identified, and the source audio signal can be selectively equalized to reduce an amplitude of the source audio signal at a frequency that correlates to the distortion signal.

Abstract: Method and apparatus for microphone matching for wearable directional hearing assistance devices are provided. An embodiment includes a method for matching at least a first microphone to a second microphone, using a user's voice from the user's mouth. The user's voice is processed as received by at least one microphone to determine a frequency profile associated with voice of the user. Intervals are detected where the user is speaking using the frequency profile. Variations in microphone reception between the first microphone and the second microphone are adaptively canceled during the intervals and when the first microphone and second microphone are in relatively constant spatial position with respect to the user's mouth.

Abstract: A signal processor for use in a vehicle audio system provides multiple operating modes. A first mode provides substantially equal total sound pressure levels at each seating location. A first equalization pattern provides a substantially similar frequency response at each of the seating locations, and a first balance configuration provides substantially similar balance patterns at each of the seating locations. A second mode provides greater sound pressure levels at a first seating location than at other seating locations. A second equalization pattern results in a frequency response at the first seating location that is substantially smoother than the frequency responses at other seating locations, and a second balance configuration results in the balance pattern at the first seating location being substantially more balanced than the balance patterns at other seating locations.

Abstract: A method of setting an equalizer so as to enlarge a sound field in reproducing an audio file and a method of reproducing an audio file thereby, includes: dividing an input audio file into segments with a predetermined time length; extracting an audio feature value for each segment; determining equalizer information for reproducing each segment by the use of the extracted feature value; and determining an equalizer sequence for the audio file by the use of the determined equalizer information of each segment. Since the equalizer setting information can be automatically changed without user manipulation, the user can listen to an audio file of which the sound field is dynamically enlarged.

Abstract: A system and method for graphic equalization of audio signals is disclosed. Traditional graphic equalizers provide control over the gains in each of a set of frequency bands. However, the actual band gains vary from the desired gains due to crosstalk between bands. Prior art methods for addressing this difficulty include applying a correction filter to the equalizer, and adjusting the shape of the individual band filters, both of which increase the computational cost. In an embodiment of the present invention, the input gains are processed to produce a set of adjusted gains which take into account the crosstalk, and result in an equalization interpolating the input band gains.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: An audio compensation unit includes a detection subunit and a compensation subunit. The detection subunit is for detecting an audio signal to obtain an original frequency. The detection subunit defines a predetermined low frequency band and a predetermined high frequency band. The detection subunit is also for generating a first control signal when the original frequency is within the low frequency band, and a second control signal when the original frequency is within the high frequency band. The compensation subunit includes an electronic switch, a first compensation circuit, and a second compensation circuit. The electronic switch enables the first compensation circuit according to the first control signal, for compensating the audio signal. The electronic switch is further enables the second compensation circuit according to the second control signal, for compensating the audio signal.

Abstract: An audio processor for generating an audio output signal (Y) with an enhanced spectral component compared to an audio input signal (X). The processor comprises a frequency splitter (FS) for splitting the input (X) into first and second parts (X1, X2) representing different frequency bands. A gain calculator (GC) estimates a level (LY) of the enhanced audio output signal (Y) for a case where the first signal part (X1) is gained by a previous enhancement gain value (LG). A dynamic headroom (HR) available is calculated based on the estimated level (LY). An enhancement gain (G) is then calculated based on the available dynamic headroom (HR), and this enhancement gain G is applied to the first signal part (X1), and finally the enhanced output signal (Y) is generate by combining the enhanced signal part (ESP) and the second signal part (X2). Preferably, the enhancement gain G calculation is updated for each signal sample, thus allowing fast adjustments of the gain G to avoid clipping distortion.

Abstract: Electronics for altering the audio frequency response of a micro-speaker without modifying the micro-speaker itself; the micro-speaker having a resonant peak region. In one embodiment the electronics includes a first circuit for flattening the frequency response curve up to the resonant peak region, and a second circuit for flattening the frequency response curve for audio frequencies higher than this region. Preferably, the extent of the flattened response over such range of frequencies is in the range of plus or minus 3 dB. The first circuit includes one of the group consisting of a high pass filter and a low pass filter, while the second circuit includes the other of this group. Each filter yields an integer multiple of 6 dB per octave slope. In another embodiment, for correcting hearing loss, a high pass filter is connected to the micro-speaker to progressively attenuate the frequency response curve as the frequency decreases.

Abstract: A sound system (8) comprises a transducer unit (1) and a frequency mapping device (2). The transducer unit comprises an array of transducers (14). The frequency mapping device is arranged for mapping a frequency range of an audio input signal (Vin) onto a frequency at which the transducer unit has a maximum efficiency, such as a resonance frequency or the Helmholtz frequency of the transducer unit. The transducer unit (1) may comprise a dedicated transducer (13) is coupled to the frequency mapping device (2) for producing the Helmholtz frequency of the transducer unit (1). The sound system may additionally comprise a sound processing device (3) arranged for steering sound produced by the array of transducers (14).

Abstract: Disclosed herein is a recording apparatus including: an audio signal correction block configured to execute correction for flattening the frequency characteristic of an audio signal supplied from a microphone and/or correction of the level of the audio signal; a correction control block configured to make the audio signal correction block adjust the level of the audio signal attenuate the reference value with time, and, if an audio signal with a level thereof exceeding the reference value is entered, use the absolute value of the level of the audio signal exceeding the reference value as a new reference value; and a recording block configured to record the audio signal to a recording media.

Abstract: A light-emission responder can respond to input of sound, realizes versatility of response, and can be applied to various use forms. A microphone (110) converts a sound wave into an electric signal. An amplifying section (111) amplifies the electric signal and outputs it to an AGC section (112). The AGC section (112) adjusts the amplitude of the electric signal output from the amplifying section (111). A filter section (113) outputs an electric signal in a frequency band f1 out of the amplitude-adjusted electric signal to a comparing section (114). The comparing section (114) compares the input electric signal with a reference signal. If the voltage of the electric signal passing through the filter section (113) is higher than the reference voltage, the comparing section (114) outputs a signal “H”. When a signal “H” is output from the comparing section (114), a drive section (115) allows a light-emitting element (120) to emit light.

Abstract: Suppressing one or more frequency ranges of a signal prevents the occurrence of feedback in a voice data communications application. A system recognizes a frequency range in a signal where feedback occurs, or anticipates a frequency range where feedback is anticipated. The signal includes a signal the input system generates or that the output system renders. The system suppresses the signal in the frequency range by disregarding one or more sampling bits representing the frequency range, or by applying one or more filters to attenuate or eliminate the signal in the frequency range. The system may monitor the signal to identify feedback resulting in different or additional frequency ranges and suppress the signal in the different or additional frequency ranges to prevent feedback from occurring.

Abstract: A method of generating a test tone signal includes generating a fundamental tone signal, which is a sinusoidal signal having a predetermined frequency; generating a first group of harmonic tone signals having different frequencies that are integral multiples of the predetermined frequency; generating a second group of the harmonic tone signals having different frequencies that are integral multiples of the predetermined frequency, at least part of the second group of the harmonic tone signals having frequencies different from frequencies of the first group of the harmonic tone signals; adding the fundamental tone signal to the first group of the harmonic tone signals to generate a first test tone signal; adding the fundamental tone signal to the second group of the harmonic tone signals to generate a second test tone signal; and outputting the first and second test tone signals at predetermined intervals.

Abstract: In a user terminal, a reproducing unit reads out from a music file memory unit a music file selected and input by a user and reproduces the music file. In a normal reproduction, an output unit outputs the reproduced sound as an audio. When a notification information reception unit receives information that needs to be noticed such as an e-mail or a phone call, a notification information analysis unit analyzes the setting of the information or notification sound and determines the degree of importance. A frequency band allocation unit selects from allocation patterns of frequency bands stored in an allocation information memory unit a pattern for the music and for the notification sound according to the degree of importance and the frequency band of the notification sound. An audio processing unit extracts and synthesizes the components of the frequency bands allocated to music and the notification sound, and the output unit outputs accordingly.

Abstract: The invention features a dual-mode audio system for an automobile which in one mode is configured for play while the doors to the automobile are closed, and in a second mode, is configured for play while a door, such as a tailgate, is opened. Additionally, the invention also features a battery monitor which monitors the operating condition of the battery and may estimate the energy capacity of the battery. The system may also take power conservation measures if the system determines that the engine is not running or the operating condition of the batter reaches a predetermined state. The invention also features a remote control integrated within a key fob for controlling the audio system. The invention features various orientations for mounting a pair of loudspeakers within an automobile which directly radiate sound out of a rear opening of the automobile.

Abstract: The present invention a parametric equalizer and a method for correcting for room mode problems. The inventive device is an analog based, parametric notch filtration system with at least two-channels. The inventive equalizer has sufficient bands, bandwidth, attenuation, and flexibility to accurately address room mode problems. The inventive system allows for continuous fine adjustment of attenuation and bandwidth. The method involves measuring, evaluating, and calculating what variables need to be used to correct the room mode issue for a particular room and then correcting the identified variables.

Abstract: Graphic equalizer as a cascade of equalization filters with the gain of each individual filter compensated for the gain leakage from other filters. A linear approximation allows individual filter gains to be set to give desired frequency responses.

Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.

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: A signal processing device includes a bass signal extractor, a harmonic wave generator, a level detector, and an adjustment controller. The bass signal extractor first extracts a bass signal from an input audio signal. Natural-sounding bass enhancement is achieved as a result of the adjustment controller boosting the bass signal level until the level detector detects the bass signal level at a set level. For input bass signal levels higher than the set level, bass is enhanced virtually using a harmonic signal generated from the bass signal by the harmonic wave generator. As a result, the disadvantages of the boost method and the virtual signal enhancement method are mutually compensated for, and synergistic advantages for bass enhancement are obtained.

Abstract: Loudness-based compensation systems and techniques are described which provide audio compensation in noisy environments. Loudness approximations are determined for an audio block both by itself and in the presence of background noise. These approximations utilize compression of audio intensity within frequency bands in order to better reflect listeners' hearing perception. From these loudness approximations, a gain is determined for the audio block and then applied in such a manner that the effect is not jarring to listeners.

Abstract: A computer audio system includes an audio codec and a tone controller. The audio codec is operably coupled to receive audio information, which includes tone control settings, PCM digital audio inputs and PCM digital audio outputs. In addition, the audio codec may receive audio information as analog input signals via a line-in, a CD input, or an auxiliary input. Based on the audio information, the audio codec provides a first stereo output, a second stereo output and a monotone audio output. The tone controller is operably coupled to the audio codec and includes a low pass filter, a high pass filter, a band pass filter, and a summing module. The low pass filter is operably coupled to filter the monotone audio output and isolates bass components of the audio signal being processed. By further coupling a volume control module to the low pass filter, the bass component of the audio signal being processed may be varied.

Abstract: An audio codec includes an input for receiving audio information. Audio processing circuitry produces a first stereo audio signal, a second stereo audio signal, and a monotone audio signal based on the audio information. A low pass filter filters the monotone audio output, wherein the low pass filter passes a bass component of the monotone audio signal substantially unattenuated and attenuates higher frequency components of the monotone audio signal. A high pass filter filters the first stereo audio output, wherein the high passes filter passes a treble component of the first stereo audio signal substantially unattenuated and attenuates lower frequency components of the first stereo audio signal. A band pass filter filters the second stereo audio output, wherein the band pass filter passes a mid band component of the second audio signal substantially unattenuated and attenuates low frequency components and high frequency components of the second stereo audio signal.

Abstract: An audio-signal-processing apparatus includes a band-decomposition unit utilizing a band-decomposition characteristic based on the harmonic-tone composition of musical instruments. A low interval limit generates the harmonic-tone to each band signal after decomposing the bass signal into a plurality of band signals. The audio-signal processing apparatus generates a harmonic-tone individually for each pure tone component, which includes complex tones and suppresses deterioration of the sound quality.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: The invention relates to a filter circuit for processing an audio signal, comprising a first branch in which the audio signal is guided in an essentially unchanged state to an output summation stage, and a second branch comprising several series-connected filter stages. The audio signal is inputted into a first filter stage of the series-connected filter stages and guided via said stages to an output summation stage. The second branch comprises several secondary branches via which the audio signal is directly guided to further filter stages of the series-connected filter stages. At least one of the filter stages is adjustable and adjusting one filter stage has an impact on the function of another filter stage.

Abstract: A computer audio system includes an audio codec and a tone controller. The audio codec is operably coupled to receive audio information, which includes tone control settings, PCM digital audio inputs and PCM digital audio outputs. In addition, the audio codec may receive audio information as analog input signals via a line-in, a CD input, or an auxiliary input. Based on the audio information, the audio codec provides a first stereo output, a second stereo output and a monotone audio output. The tone controller is operably coupled to the audio codec and includes a low pass filter, a high pass filter, a band pass filter, and a summing module. The low pass filter is operably coupled to filter the monotone audio output and isolates bass components of the audio signal being processed. By further coupling a volume control module to the low pass filter, the bass component of the audio signal being processed may be varied.

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 and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.