Abstract: The bass compensation device of the present invention includes a high frequency removing circuit for removing a high frequency component of an input sound signal; an absolute value circuit for full-wave rectifying an output signal of the high frequency removing circuit; a band limit circuit for selectively passing only a predetermined frequency range component from the output signal of the absolute value circuit; and an amplifier for amplifying the output signal of the band limit circuit, and bass compensation is performed by the output signal of the amplifier. This structure can realize a bass compensation device excellent in distortion-factor property and transient property.

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: An audio tone-control circuit includes 4th-order low-pass, 4th-order band-pass, and 4th-order high-pass filter circuits. The 4th-order low-pass filter circuit has a first phase response and receives and filters an audio signal and generates a first output signal therefrom. The 4th-order band-pass filter circuit has a second phase response that is substantially equal to the first phase response and receives and filters the audio signal to generate a second output signal. The 4th-order high-pass filter circuit has a third phase response substantially equal to the first and second phase responses and receives and filters the audio signal to generate a third output signal. The audio-signal circuit also includes a combining circuit that combines the first, second and third output signals into a combined signal. Furthermore, the filter circuits may be 2nd-order filter circuits instead of 4th-order filter circuits.

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: The present invention relates to an auto loudness circuit for performing loudness compensation automatically depending on the signal level. When the signal level decreases, loudness compensation is slowly introduced and as the signal level increases, loudness compensation is quickly removed. To do so, the auto loudness circuit utilizes a filter circuit with the characteristic of a first order bass boost. The filter circuit maintains a corner frequency which is proportional to the inverse of audio level in order to mimic the Fletcher-Munson curves. Because the circuit employs a capacitance-multiplier with a first order resistance capacitance filter, the bass boost is inversely proportional to the signal level. Thus, bass boost is achieved automatically as the program content changes so that the listener is unaware of significant changes in program material as signal levels change either through increase or decrease in volume, crescendo or new material.

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: 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: 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 speaker includes at least one transducer and at least one memory device. The at least one transducer is adapted to receive an audio signal. The at least one memory device is adapted to store data related to the speaker. A method includes reading data from a memory device of at least one speaker. An audio signal is provided from an audio system.

Abstract: A high pass filter 32 having a cut-off frequency of 200 Hz and a cut-off response of +12 dB/OCT and a low pass filter 33 having a cut-off frequency of 400 Hz and a cut-off response steeper than ?24 dB/OCT are used to pick out components in a double overtone region of 200˜400 Hz of a bass musical instrument such as a base or a bass drum from an input audio signal. The component which is picked out is input to distortion applying means 34 having an input-output response which is a non-linear response having no point symmetry with respect to 0 point, thus applying a distortion to cause even-numbered overtone components to be produced. An output from the distortion applying means 34 and the input audio signal are summed in a summer 18 to be delivered. Natural tones which have boosted bass tones and which are free from impurity are obtained.

Abstract: Data is encoded as a number of audio tones forming a leader tone, a series of tones forming a start symbol and a greater number of data symbols following the start symbol which are each formed of a group of audio tones. The information is transmitted from a speaker or similar transducer to a microphone or similar transducer. The microphone is coupled to the input of a personal computer within which software is operative to analyze and decode the information. The leader tone is used to determine any frequency inaccuracies or errors which have occurred in the system. Based upon detected frequency inaccuracy or deviation, a frequency offset is employed within the system upon all received audio tones to adapt the system to the frequency inaccuracies or errors and produce correct data.

Abstract: Audio tracks including environmental information are recorded on a DVD, or other playback medium using microphones, vibration sensors, or other devices that capture ambient sound from the original environment in which a performance, presentation or other sound is created. For example, environmental audio tracks can be obtained by placing microphones on the floor, ceiling and walls of an amphitheater during a pop music performance. Any object, item or surface can be a candidate from, or within, which to capture an ambient sound. Sounds can be obtained from sensors mounted to musicians, audience members, theater seats, various rooms within a building, etc. The environmental audio tracks (EATs) can be provided individually or mixed together with one or more other EATs, one or more standard recorded tracks, or otherwise mixed or combined with other sounds. Or the EATs can be processed, effected, etc., as desired.

Abstract: A band-dividing unit is operable to extract a low frequency component from an input signal in order to generate overtones based on the extracted low frequency component, and is further operable to divide the extracted low frequency component into signals that belongs to different frequency bands. Each of overtone-generating units is disposed for corresponding one of the different frequency bands, and is operable to generate overtones based on an output signal from corresponding one of band pass filters. An adder adds the generated overtones to the input signal that has passed through a delay. The resulting acoustic signal is sent to the outside through a high-pass filter. One overtone-generating unit designed for a higher frequency band among the different frequency bands is set to produce the same or fewer overtones than another overtone-generating unit suited for a lower frequency band thereamong does.

Abstract: The tone generator uses a highly accurate frequency signal to generate an accurate and stable tone signal. A cadence pattern may be used to output the tone signal onto a test cable in an easily distinguishable pattern. The tone signal may be selectively output at an in-band band frequency or an out-band frequency. A corresponding tone probe may be used to detect the tone signal. The tone probe uses a high-Q bandpass filter to reject all but the expected frequency of the tone signal. The signal passing the bandpass filter is proportional to the detected tone signal. The tone probe generates an audio signal unrelated to the frequency of the tone signal. The audio signal is modulated by the signal from the bandpass filter, and then broadcast through a speaker. Accordingly, the tone probe provides an audible sound proportional to the strength of the tone signal, but is unrelated in frequency.

Abstract: A audio communication system for a vehicle includes a primary audio source producing a primary audio signal, a secondary audio source producing a secondary audio signal, a speaker, a vehicle ignition switch having an on position and an off position, and a relay. The primary and secondary audio sources may be alternatively transmissibly connected to the speaker by the relay. The relay receives the primary audio signal and transmits substantially the primary audio signal to the speaker when the vehicle ignition switch is in the on position and the secondary audio signal is not present. The relay receives the primary and secondary audio signals and transmits substantially the secondary audio signal to the speaker when the vehicle ignition switch is in the on position. The relay receives the secondary audio signal and transmits substantially the secondary audio signal to the speaker for a predetermined period of time after the vehicle ignition switch has been switched from the on position to the off position.

Abstract: System and method for estimating signal parameters (e.g., frequency, amplitude and/or phase) of one or more tones present in an input signal. Samples of the input signal are received, and a frequency transform of the samples generated. An amplitude peak in the frequency transform corresponding to the tone is identified. Two or more frequency bins are selected proximate to the identified frequency in the transform. A tone frequency value is determined that minimizes a difference between two or more expressions, each including respective numerator and denominator terms corresponding to respective frequency bins and whose ratios each represent a complex amplitude of the tone at a respective bin. Each expressions includes a tone frequency variable that represents a correct tone frequency value of the tone.

Abstract: Disclosed are a tone converter and a tone converting method employed in a process of playing music, voice, and an audio signal. The tone converter includes at least one delayer for generating an echo pulse by delaying an input audio signal for a predetermined time, a multiplier for multiplying a time delay value of the echo pulse outputted from a corresponding delayer by a gain value, and an adder for adding an echo signal outputted from the multiplier to the input audio signal so as to output a tone converted audio signal.

Abstract: An electroacoustic transducer (1) includes a ringing signal generation means (20) for generating a ringing signal (CS) and an electroacoustic transducer (9) for the acoustic reproduction of the ringing signal (CS), in which the ringing signal generation means (20) includes at least two fundamental tone signal generation stages (GT1, GT2, . . . GTn), which are each adapted to generate a fundamental tone signal (GS1, GS2, . . . GSn) of a given frequency (f1, f2, . . . fn), which frequencies (f1, f2 . . . fn) are in a rational ratio to each other, and in which the ringing signal generation means (20) has a control means (21) with the aid of which each of the fundamental tone signal generation stages (GT1, GT2, . . . GTn) can be activated for a given time interval, namely in such a manner that at least two fundamental tone signal generation stages (GT1, GT2, . . . GTn) are activated simultaneously.

Abstract: Audio produced by an electronic appliance is selectively filtered based upon the output device selected. The destination of the audio signal is detected and filtering of the audio signal is configured based on the destination. If the frequency response of the output device is narrow, such as with an internal speaker in the electronic appliance, a bandpass filter is applied to narrow the frequency range of the audio signal to match the frequency response of the output device.

Abstract: A method and apparatus for converting the reproducing speed of an acoustic signal where, of the acoustic signals held in a data recording section 1, the input acoustic signal s1 (sampled for max. pitch cycle×2) is read from a process-start position P. A low-pass filter 7 controls the high-band component of the acoustic signal s1. A decimation section 8 performs appropriate down-sampling on a signal output from the low pass filter 7. The signal, thus down-sampled, is read into a signal buffer section 9. A down-sampled, input acoustic signal s2 is transferred from the signal buffer section 9 to a pitch-calculating section 3, which calculates a pitch cycle s3.

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 unique, fully integrated, fully programmable, and highly flexible sound distribution system and methodology for providing masking sound, background music, and paging capabilities in up to eight zones of a building or space is provided. The methodology embodied in the system includes internal masking sounds that are uniquely pre-filtered to provide efficient and effective masking of distracting sounds within selectable zones of the space with a minimum masking sound dB sound level and with a pleasant sounding and non-annoying masking sound. The system also incorporates the capacity to be controlled from a remote or local telephone to adjust the volume level in any zone serviced by the system by issuing appropriate DTMF codes from the telephone's keypad. Unique bi-tone diagnostic functions are provided for assuring that the entire system is correctly wired and installed and for troubleshooting operational anomalies.

Abstract: A sound field forming circuit includes a first switch portion which switchingly supplies one of an audio signals from a sound source and a second tone control circuit which variably adjusts a characteristic of a second predetermined frequency band, to a first tone control circuit which variably adjusts a characteristic of a first predetermined frequency band; a second switch portion which switchingly supplies one of the audio signals from the sound source and the first tone control circuit, to the second tone control circuit; a third switch portion which switchingly supplies one of the audio signals from the first tone control circuit and the second tone control circuit, to a front channel; and a fourth switch portion which switchingly supplies one of the audio signals from the first tone control circuit and the second tone control circuit, to a rear channel.

Abstract: A sound signal reproducing apparatus for supplying sound signals for a plurality of channels to speakers via sound amplifier circuits includes: power supply voltage control means for controlling power supply voltage to the sound amplifier circuits; a plurality of filter circuits for passing a signal of 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 level of output signals of the plurality of filter circuits; an adder circuit for adding together 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 control means.

Abstract: A digital base booster (DBB) for reducing hardware by using an arithmetic processor is provided. Instead of using a conventional IIR filter having a cascade structure including a plurality of partial building blocks, the digital base booster using an arithmetic processor includes first internal data, an inputting portion, a data assigner, an arithmetic portion, and an output data storing device. The first internal data is the output data of the arithmetic portion. The inputting portion includes a plurality of multi-bit registers, thereby storing input data and the first internal data and outputting the stored data in a predetermined signal. The data assigner selects one output data from a plurality of output data of the inputting portion. The arithmetic portion performs an arithmetic operation on the output data of the data assigner and data stored in the arithmetic portion, compensates for and stores a round-off error of the data output by the operation, and outputs the first internal data.

Abstract: To encode a URL in sound, the characters of the URL are mapped to sound codewords each of which is used to produce, in a sound output, a sound feature particular to that codeword, the nature of the sound features and of the overall mapping between characters and sound features being such that at least certain character combinations that occur frequently in URLs produce sound sequences of a musical character. Decoding of the sound URL effects the reverse mapping.

Abstract: An invention to off-load the processing requirements for the playback of audio sequences within an electronic device. A dual processing system is utilized in which a host processing unit manages a plurality of tone patterns and detects the occurrence of events that require the playback of one or more of the tone patterns. The host processing unit transfers the appropriate tone patterns to a supplemental processing unit. The supplemental processing unit performs all of the necessary conversions for transforming the tone patterns into audio sequences and playing the tone sequences. This advantageously allows the host processing unit to process time-critical events without being burdened by providing the playback of the tone patterns.

Abstract: Multi channel audio reproduction has already been used for some time. For example the so-called surround sound signals can improve the appearance of reproduced sound to a great extent.

Abstract: A sound field forming circuit includes a first switch portion which switchingly supplies one of an audio signals from a sound source and a second tone control circuit which variably adjusts a characteristic of a second predetermined frequency band, to a first tone control circuit which variably adjusts a characteristic of a first predetermined frequency band; a second switch portion which switchingly supplies one of the audio signals from the sound source and the first tone control circuit, to the second tone control circuit; a third switch portion which switchingly supplies one of the audio signals from the first tone control circuit and the second tone control circuit, to a front channel; and a fourth switch portion which switchingly supplies one of the audio signals from the first tone control circuit and the second tone control circuit, to a rear channel.

Abstract: An audio system that automatically corrects for variations in spectral balance in audio source material. The audio system comprises a power spectrum analyzer and a source correction equalizer unit coupled to receive an audio input signal, and an index control unit coupled to a spectral balance correction data memory. A reference spectral balance may initially be established by playing a reference source medium with desired sound attributes. The power spectrum analyzer characterizes the spectral balance of the reference source medium, storing the results in non-volatile memory. When a particular source medium is played for the first time, the index control unit reads an identifying parameter from the particular source medium. Because there is no entry for the identifying parameter in spectral balance correction data memory, the particular source medium has not been previously played on the audio system.

Abstract: The invention relates to a method of providing a user with information on the operation of the portable device and to a portable device. In the device, such a tone is produced that, due to a tone feature, can be distinguished from background noise. This feature can be tone frequency, duration, volume or moment of time. The device can analyse background noise automatically, and based on this, it adjusts at least one feature of the tone automatically such that the tone can be distinguished from background noise, and the background noise does not mask out the tone. Alternatively, the user himself can adjust the tone frequency or duration in a desired way so that it would be distinguished from background noise more clearly.

Abstract: A pitch shifter capable of shifting an acoustic signal in pitch to an arbitrary level with a high degree of accuracy without any change in reproduction time, and also sufficiently reducing high-frequency distortion without being increased in size or speeded-up is provided. Stored in a filter coefficient string storage 6, four filter coefficient strings corresponding to four sub-filters produced through polyphase decomposition of a low-pass filter for 4-fold oversampling. Filter coefficient string selectors 5a and 5b select, based on the first and second bits of the decimal part of each of read addresses generated by the read address generators 4a and 4b, respectively, any one of the four filter coefficient strings stored in the filter coefficient string storage 6. Filter operation units 2a and 2b receive paired sound data strings, and carry out a filter operation by using the filter coefficient strings selected by the filter coefficient string selector 5a and 5b, respectively.

Abstract: A pitch shift apparatus is provided to pitch shift a digital audio signal into a pitch-shifted signal. The apparatus comprises a receiving means, a pitch shifting means and a connecting means, wherein the connecting means comprises: a search region comparator for comparing each sample in the search region with a reference level to obtain a search region bit sequence representing the amplitude of each sample in the search region; a cross region comparator for comparing each sample in the cross region with the reference level to obtain a cross region bit sequence representing the amplitude of each sample in the cross region; a bit processor for bit comparing the cross region bit sequence and any sub-search region bit sequence of M samples in the search region to obtain a corresponding non-similarity; and a connecting device connecting the cross region and a sub-search region corresponding to the minimum non-similarity to renew the pitch-shifted signal.

Abstract: Again control arrangement for an auditory prosthesis, such as a hearing aid or cochlear prosthesis, is disclosed. The output of a controlled amplifier 11 is processed so as to derive an average noise floor level. This is compared to a predetermined level, and the gain adjusted slowly up or down in response to the comparison. Hardware and software implementations are disclosed.

Abstract: A tuner is presented wherein each o a plurality of RF AGC gain controllable amplifiers are individually controlled by individual AGC control signals generated by an AGC controller so that the level of the output signal from each of the RF AGC gain controllable amplifiers is individually optimized for tuner performance.

Abstract: A volume/tone control circuit comprises first and second latch circuits (8) and (9) for latching the volume control data BD and the tone control data TD stored in the shift register (7); a volume regulating circuit (2) and a tone regulating circuit (3) for regulating the volume and the tone of the input audio signal according to the respective output data of the first and second latch circuits (8) and (9); and a zero-crossing detection circuit (12) for detecting zero-crossings of the input signal. The input audio signal is fed to the zero-crossing detection circuit (12) when a data detection circuit (20) detects that the input data and the output data of the first latch circuit (8) differ.

Abstract: An audio signal processor is disclosed which has at least one audio signal input and at least one audio signal output as well as at least one control input, and an audio signal processing unit connected between audio signal input and audio signal output, the audio signal processing unit having a programmable indicator tone generator circuit to be driven via the control input and whose indicator tone signal is switched to the audio signal output in accordance with the state of the control input.

Abstract: An equalizer includes first and second differential amplifiers which have a common output and respective separate current sources, a first negative feed back circuit which feeds back a voltage signal extracted from the common output to the first differential amplifier, a second negative feed back circuit which feeds back the voltage signal to the second differential amplifier, and a filter circuit which is provided at either the first or the second negative feed back circuit and which differentiates frequency characteristics during signal amplification in the first and second differential amplifiers. Frequency characterstic is selected through selection of one of the current values of the respective current sources in response to a sound quality adjusting signal.

Abstract: An arrangement for reproducing audio signals includes a pitch control device for increasing or decreasing the amplification of audio signals in one or more defined frequency ranges and a volume control device for increasing and decreasing the amplification of the audio signals in a full effective frequency range of the audio signals wherein the pitch control device reduces the amplification in the defined frequency ranges if the amplification of the audio signals in the full effective frequency range is increased by the volume control device and wherein the pitch control device increases the amplification of the audio signals in the defined frequency ranges if the volume control device decreases the amplification of the audio signals in the full effective frequency range.

Abstract: A power amplifier device for amplifying an audio signal from sound generating circuitry in a micro-computer is provided. The device is retained in a housing secured in a drive bay of the micro-computer containing a disk drive such as a compact disc read only memory drive, a compact disc recordable drive, a digital video disc drive, etc. The amplifier device and disk drive may also be configured in a unitary housing. The device receives a line-level audio signal from the audio circuitry and transmits the signal to a frequency and time response contouring device, to adjust the response of the signal. A power level adjusting mechanism, such as a volume control, adjusts the power level of the signal generated by the frequency and time response device. Amplifiers are coupled to the volume control to amplify the audio signal to a level determined by the volume control. The amplification device connects to at least leftwardly and rightwardly positioned loudspeakers to provide stereophonic sound at the desired level.

Abstract: An equalizer includes first and second differential amplifiers which have a common output and respective separate current sources, a first negative feed back circuit which feeds back a voltage signal extracted from the common output to the first differential amplifier, a second negative feed back circuit which feeds back the voltage signal to the second differential amplifier, and a filter circuit which is provided at either the first or the second negative feed back circuit and which differentiates frequency characteristics during signal amplification in the first and second differential amplifiers. A frequency characteristic is selected through selection of one of the current values of the respective current sources in response to a sound quality adjusting signal.

Abstract: A power amplifier device for amplifying an audio signal from sound generating circuitry in a micro-computer is provided. The device is retained in a housing secured in a drive bay of the micro-computer containing a compact disc read only memory drive. The device and disc drive may also be configured in a unitary housing. The device receives a line-level audio signal from the audio circuitry and transmits the signal to a frequency and time response contouring device, to adjust the response of the signal. A power level adjusting mechanism, such as a volume control, adjusts the power level of the signal generated by the frequency and time response device. Amplifiers are coupled to the volume control to amplify the audio signal to a level determined by the volume control. The amplification device connects to at least leftwardly and rightwardly positioned loudspeakers to provide stereophonic sound at the desired level.

Abstract: A multi-band equalizer for receiving and processing an audio signal includes a plurality of operational amplifiers; each for processing a different frequency band of the audio signal to produce an output signal. Each operational amplifier includes an inverting input, a non-inverting input, a resistor means coupled to the non-inverting input for receiving the audio signal and supplying it to the non-inverting input, an output, a feedback circuit coupled between the output and the inverting input for determining the upper limits of the frequency band processed by that particular operational amplifier, and an input circuit coupled between the inverting input and ground potential for determining the lower limits of the frequency band processed by that operational amplifier. A summing circuit is coupled to the outputs of all of the operational amplifiers for combining the output signals thereof to produce a resultant signal.

Abstract: An apparatus for automatically recalling audio parameter setup is disclosed. A processor is coupled between a MIDI interface and one or more analog level controlled audio processor channels. The setup parameters, previously entered and stored by the composer in a host computer, are transmitted via the MIDI interface to the processor. The parameters are subsequently converted into an analog signal by a converter. The analog signal is provided to a parameter conversion array which converts the analog signal based on a transformation function. The outputs from the parameter conversion array are provided to one or more analog multiplexers. The outputs of the multiplexers are provided to the control inputs of the analog signal processing channels. Each control input of each analog signal processing channel includes a small capacitor, which in combination with an operational amplifier, forms a sample-and-hold circuit to temporarily store the analog output for the analog processor channels.

Abstract: A recording and reproducing apparatus for an audio signal including a low pass filter removing high frequency components of a predetermined frequency or more of an audio signal having a wide band, an A/D conversion circuit converting the audio signal from the low pass filter into a digital signal, a memory for storing in order to record a predetermined period of time of the digitized audio signal from the A/D conversion circuit, a D/A conversion circuit converting the audio signal read from the memory into an analog signal, a tone control circuit conducting a frequency correction on the audio signal, and a control circuit controlling the tone control circuit such that, with respect to the audio signal recorded in the memory or the audio signal read from the memory low frequency components of the audio signal to be reproduced are attenuated in correspondence with the high frequency components of the audio signal which have been removed by the low pass filter whereby the audio signal with no conspicuous unnatur

Abstract: An equalizer, for providing a tunable pole/zero compensated output signal, includes a integrated tone control circuit with a MOSFET-C configuration such that the spacing between the pole and the zero and the center frequency of the pole/zero pair can be tuned by varying the resistance of MOSFET resistors using variable voltage sources applied to the gates of the MOSFETs.

Abstract: A method is disclosed of modification of parameters of audio signals by dividing a digital signal converted from an original analog signal into sound frames, modifying a pitch and a playing rate of the digital signal within a frame and subsequent successive splicing a last modified frame with a first non-modified frame and calculating a differential mean absolute error to define the best splicing point in terms of producing minimal or no audible noise such that various sections of sound signals can be spliced together to achieve pitch and playing rate modification. An apparatus is also disclosed for implementing the method, the apparatus comprising input and output amplifiers, a low pass filter at the input and a low pass filter at the output, analog-to-digital and digital-to-analog converters, and a pitch shifting processor.

Abstract: A system for automatically adapting the mean sound level of a television receiver. The system for automatically adapting the mean sound level of a television receiver comprises an adapting module comprising both a video adapting circuit comprising at least one additional input highpass filter and a fixed-gain video amplifier connected in series and interposed in the video image processing circuits at circuity for providing video image signals, and secondly an audio adapting circuit interposed in the sound path at circuity for taking audio signals to detect the input mean level of sound, to compare said mean level of sound with a pre-established reference mean level, and automatically to adapt the mean level of output sound to maintain it at the pro-established reference mean level.