Abstract: This specification describes technologies relating to reducing audio masking. In general, one aspect of the subject matter described in this specification can be embodied in methods that include the actions of receiving a primary audio signal and a secondary audio signal; for each audio signal, calculating an average perceived intensity over time for each of a plurality of frequency bands; comparing the average perceived intensity of the secondary audio signal with the average perceived intensity of the primary audio signal for each frequency band; and for each frequency band where the average perceived intensity of the secondary audio signal is greater than the average perceived intensity of the primary audio signal by a specified threshold amount, attenuating the secondary audio signal by a specified amount to form a modified secondary audio signal.

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: 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: An automatic gain control unit controls the gain applied to an input signal produced by a microphone subject to ambient noise. The automatic gain control circuit continually monitors the signal level of said input signal. A first gain control circuit decreases the gain applied to the input signal in increments of a first size when the input signal exceeds a first predetermined level. A second gain control circuit increases the gain applied to said input signal in increments of a smaller size when the input signal falls below a second predetermined level and in response to the presence of a speech present signal. The second predetermined level is less than said first predetermined level. In one embodiment, the first gain control circuit controls the analog gain of a codec and the second gain control circuit controls the digital gain of the codec.

Abstract: There is described a method for operation of a hearing device system with two microphones arranged spatially separated from one another and with sound-generating output units assigned to these microphones, in which, by comparison of the microphone signals or of signals derived therefrom, feedback is detected, and on detection of the feedback measures are initiated for reducing the feedback, and with the comparison of the microphone signals or of signals derived therefrom comprising a frequency-selective power comparison. There is also described a hearing device system suitable for this method.

Abstract: A noise extraction method in which an environmental input which includes a noise indicia is selectively modified in accordance with an algorithm that includes one or more factors representing time response, amplitude of response, and error correction. The algorithm may also include thresholding, delay or convergence, among other techniques.

Abstract: An audio amplifier such as for driving headphones. The amplifier includes multiple amplifier devices coupled in parallel. Both a bias generator and a volume control are responsive to a user setting. Under low output signal conditions, one or more of the amplifier devices are disabled in response to the user setting. Disabled amplifier devices do not consume output bias current. Thus the audio amplifier has reduced power consumption, and the system has longer battery life.

Abstract: A system automatically determines an equalizing filter characteristic for a communication system within a vehicle. The communication system includes a loudspeaker and a microphone or microphone array. The system transmits a predetermined test signal through the loudspeaker and receive the test signal through the microphone or microphone array. Based on the predetermined test signal and the received test signal, a transfer function is developed. The equalizing filter characteristic is then developed from the transfer function.

Abstract: A method and apparatus for automatically adjusting volume of an audio signal on a mobile device, comprising playing the audio signal at an initial volume, sampling the audio signal, estimating a transfer function based on an echo path characteristic between the played audio signal and sampled audio signal, selecting a volume policy based on the estimated transfer function, and adjusting the volume of the audio signal in accordance with the selected volume policy.

Abstract: Methods and systems for processing audio signals are disclosed herein. Aspects of the method may comprise controlling gain during multipath, multi-rate audio processing by generating a digital signal that is a product of an input digital signal and a gain coefficient derived from a lookup table, and coarse tuning the gain by bit-shifting the digital signal to generate a digital output signal. The gain may be fine-tuned utilizing a variable step size determined by interpolation. The gain coefficient may be partitioned into gain blocks, which may be twice a corresponding value in preceding gain blocks. The gain blocks may be partitioned into steps that represent particular gain values within a range associated with the gain block. The digital output signal may be ramped utilizing a linear interpolation of the gain coefficients over a number of samples of the input signal, where the number of samples is given as a power of two.

Abstract: A front-end processing circuit applies predetermined processing to an input signal and outputs a processed signal. A frequency modulation circuit applies frequency modulation to the output signal of the front-end processing circuit and outputs a processed signal. The front-end processing circuit includes a variable amplifier and a level detector. The variable amplifier receives a feedback signal and amplifies a signal by an amplification factor corresponding to the feedback signal. The level detector outputs the feedback signal to the variable amplifier in accordance with a level of a signal obtainable in a circuit succeeding the variable amplifier. The front-end processing circuit maintains the signal supplied to the frequency modulation circuit at a constant level.

Abstract: A control system comprising a computing device, a memory device comprising a database, and a plurality of audio sources. The memory device is coupled to the computing device. The database comprises input data. The computing device is adapted to control an audio level for each of said audio sources in response to said input data to resolve audio conflicts between the audio sources.

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 a higher frequency portion of the input signal to emphasize attack transients occurring within the input signal.

Abstract: The present invention relates to a method of controlling a volume of a reproducing apparatus and a reproducing apparatus using the same, wherein a reproduction volume of an audio file can be automatically controlled in accordance with information on a volume level of the audio file. The reproducing apparatus of the present invention includes a memory for storing at least one audio file and volume level information for controlling a reproduction volume level of the audio file, and a control unit for controlling a reproduction volume of the audio file in accordance with the volume level information recorded corresponding to the audio file when the audio file stored in the memory is reproduced.

Abstract: A speaker system for a musical instrument that detects the displacement of a voice coil of a speaker and provides feedback processing. The speaker system has a preamp that alters the frequency characteristics of the an electrical signal that has been input to an input terminal, and a power amplifier that amplifies the electrical signal. A speaker is driven by the power amplifier and a feedback unit detects the displacement of the speaker and provides a feedback signal to the power amplifier. The power amplifier amplifies the electrical signal in conformance with the output of the preamp and the feedback signal.

Abstract: An audio data processing device including: a first processor; and a second processor which is connected to the first processor wherein the first processor includes: an audio data acquisition which acquires audio data of digital data; an omitting section which omits a bit corresponding to low volume which is hard to be heard by human ears from the audio data; and a transmitter which transmits the audio data in which the bit corresponding to the low volume is omitted by the omitting section from the first processor to the second processor; wherein the second processor includes: a receiver which receives the audio data transmitted from the first processor; and a reproduction data generator which generates audio reproduction data necessary to reproduce the audio data based on the received audio data.

Abstract: A sound reproducing system for use with multiple rooms includes a main control unit, and at least one secondary control unit. Thus, the respective automatic level control circuit maintains the voice signal output at a constant level, the respective power amplification circuit of the main control unit converts the voice signal into a larger voice signal, the volume control circuit of the respective secondary control unit converts the larger voice signal and the noise produced during transmission into a smaller voice signal, and the power amplification circuit of the respective secondary control unit converts the smaller voice signal into a larger voice signal output, thereby producing a voice signal output with a higher quality.

Abstract: A system for management and control of audio-related equipment. The system is a common architecture that provides the capability to manage, monitor and control all of the audio-related equipment in an audio system. To facilitate messaging between audio-related devices, each of the audio-related devices may be assigned a node ID a sequential manner as each audio-related device powers up. The system may also include a method for discovery of audio-related devices on an audio network. In some examples, a PEM may be located geographically remote from the audio-related devices. Accordingly, error logging, alarming and system operation may be monitored to provide troubleshooting from a remote location.

Abstract: An audio system includes an automatic level controller receiving at least one input audio signal from at least one audio medium. The automatic level controller includes a gain adjust module for producing an output audio signal based on the at least one input audio signal. A difference between a maximum sound level and a minimum sound level of the at least one input audio signal is greater than a difference between a maximum sound level and a minimum sound level of the output audio signal. A filter arrangement is communicatively coupled to the automatic level controller. The filter arrangement receives the output audio signal and divides the output signal into a plurality of filtered signals. Each of the filtered signals corresponds to a respective frequency band. A multiband dynamics controller is communicatively coupled to the filter arrangement.

Abstract: A satisfactory sound volume sensation from a hearing point of view can be obtained, and an effective sound volume correction operation can be obtained with an algorithm as simple as possible. Correction step 0 (correction function off, level correction value 0) to correction step 16 (level correction value—16 dB) are defined. When a first condition that is defined in such a manner as to correspond to a state in which distortion occurs is satisfied, the correction step is made to proceed by one step from the current correction step (?1 dB is added to a level correction value). When a second condition that is defined in such a manner as to correspond to a state in which distortion does not occur is satisfied, the correction step is made to return by one step from the current correction step (?1 dB is subtracted from the level correction value).

Abstract: Techniques and apparatus for limiting a sound pressure level produced by an audio output device driven by an audio player. During play, the audio player monitors the current sound pressure level of the audio output device, and in response thereto limits the sound pressure level by controlling an output level of audio signals it generates so as to reduce hearing loss caused by excess volume.

Abstract: A circuit for processing broadcast signals that includes circuitry for receiving and processing broadcast signals which contain audio information and providing a first audio signal, and circuitry for controlling the amplitude of a received second audio signal in response to a first control signal, and providing a third audio signal wherein the circuit further comprises circuitry that receives the first audio signal and provides the second audio signal for automatically limiting the amplitude of the first audio signal in response to at least one reference signal.

Abstract: A control system and associated method resolving audio conflicts. The control system comprises a computing device, a memory device comprising a database, and a plurality of audio sources. The memory device is coupled to the computing device. The database comprises input data. The computing device is adapted to control an audio level for each of said audio sources in response to said input data to resolve audio conflicts between the audio sources.

Abstract: A method for audio distortion processing is provided, whereby narrowband and wideband gains are incrementally reduced and recovered for controlling audio distortion of an audio reproduction system. Reduction limits are determined for both the narrowband and wideband gains, as a function of operator controlled inputs. If clipping is detected, the narrowband gain is initially reduced until either the narrowband gain reduction limit has been achieved or the clipping desists. The wideband gain is subsequently reduced if the reduction limit of the narrowband gain has been achieved and the clipping persists. The wideband gain is reduced until either the wideband gain reduction limit has been achieved or the clipping desists. After the clipping desists, a gain recovery process ensues, whereby the wideband gain is initially recovered until at its original level and the narrowband gain is subsequently recovered until reaching its original level.

Abstract: An electro acoustic system built-in test and calibration method utilizes a built-in self-test module to send a test signal through a first circuit device to an audio transmitter, causing the audio transmitter to output a test signal, for enabling the test signal to be received by a audio receiver and then processed by a connected circuit device and converted into a feedback digital signal to the self-test module for comparing the linearity relative to the originally provided test signal so that the parameter values and conformity of circuit devices can be optimized subject to comparison result. The test and adjustment procedure is recycled for other parameter items, and a warning signal is produced when proper adjustment cannot be done.

Abstract: The controller changes the low vehicle speed region set volume according to an operational input to the volume control switch within a first volume setting range set up in advance when the volume control switch is operated in the low vehicle speed region, and changes the high vehicle speed region set volume according to an operational input to the volume control switch within a second volume setting range set up in advance when the volume control switch is operated in the high vehicle speed region. A lower limit of the second volume setting range is set higher than an upper limit of the first volume setting range. Thus, it is possible to accurately control the volume of the audio system and improve feeling of volume change.

Abstract: A voice recognition system includes an adaptive filter and a subtractor. The adaptive filter generates a simulated talk-back voice y(n) by setting a filter coefficient simulating a transfer system in which an input voice corresponding to a voice command and a talk-back voice output from a speaker are input into a microphone and by filtering a talk-back voice x(n). The subtractor extracts the input voice by subtracting the simulated talk-back voice y(n) from mixed sound input into the microphone. With this configuration, the talk-back voice is attenuated from the mixed sound including the input voice and the talk-back voice input tedinto the microphone, and then, the mixed sound is supplied to a voice recognition engine. Accordingly, the user can input his/her voice during a talk-back operation without the need to interrupt it by pressing a speech button every time the user wishes to input the voice. The voice recognition operation time can be thus reduced.

Abstract: An BTSC encoder with an improved digital filter structure substantially implemented on a single CMOS integrated circuit is described. By cascading first and second order allpass filter structures to form a higher order digital filter, such as a Cauer low pass filter, limit cycle oscillations are reduced or eliminated, word-length growth from one stage to the next is contained, and a more efficient overall filter structure and performance is obtained.

Abstract: Systems and methods are provided for controlling the volume of a motor vehicle entertainment system. One motor vehicle includes a sound system, a motor having an idle speed, and a processor. The processor is configured to determine the speed of the motor and to adjust the volume of the sound system if the determined speed is less than or equal to the idle speed. An entertainment system coupleable to a combustion motor includes means for producing sound, means for determining the speed of the combustion motor, and means for adjusting the volume of the sound producing means if the determined speed is less than or equal to a threshold speed. One method includes the steps of determining the speed of the combustion motor and adjusting the volume of the entertainment system if the determined speed is less than or equal to a threshold speed.

Abstract: Methods and apparatus for automatically mixing a plurality of audio channels by adjusting respective gains of the audio channels using a control signal based on an aggregate of input levels of respective audio signals of the audio channels; and reducing the gain of a given one of the audio channels when an input level of the audio signal of that audio channel exceeds a threshold, irrespective of whether the control signal would permit the gain to rise higher.

Abstract: A circuit for optimizing the transmission of data on a communication channel is disclosed. According to one embodiment of the invention, a circuit comprises a transmitter circuit having a programmable output characteristic and being coupled to a transmission media. The transmission media receives serial data from the transmitter circuit and couples the data to a receiver circuit by way of the transmission media. A signal quality monitor associated with the receiver circuit generates received signal quality data. Finally, a feedback path couples the received signal quality data to the transmitter circuit.

Abstract: A control system and associated method resolving audio conflicts. The control system comprises a computing device, a memory device comprising a database, and a plurality of audio sources. The memory device is coupled to the computing device. The database comprises input data. The computing device is adapted to control an audio level for each of said audio sources in response to said input data to resolve audio conflicts between the audio sources.

Abstract: Variable attack and release system and method are disclosed for dynamically modifying the various elements of the system, including modifying compander response time to dynamically adjust for changing parameters such as environmental or input signal changes, to maintain output signals within predetermined limits. The system and method include permitting accurate and quick response to noise sources having a duration which exceeds a predetermined threshold while at the same time ignoring transient or short duration environmental noise.

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: A device for decoding audio signals subjected to a noise-reduction encoding such as a Dolby-B encoding comprises a plurality of processing blocks (612, 616, 620, 624, 628, 632, 636, 640) for generating, starting from an input signal (610) containing the audio signal subjected to encoding superimposed on a noise component, an output signal (626) consisting of a replication of the audio signal with the noise component reduced. The aforesaid processing blocks are implemented in a digital form and comprise a sliding-band filtering structure (612) fed with the input signal (610) and designed to generate a filtered signal (614). The filtered signal (614) is fed, according to a general feedforward scheme, to an overshoot-suppression stage (616) in view of the generation, in an adder node (620), of a difference signal (622) starting from which the output signal is obtained via filtering (624).

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: A circuit and related method for digital volume control are provided, where the circuit includes a digital filter configured to process samples of an input stream in a manner that processes a previous input sample during a time interval before a subsequent input sample, and outputs a series of exponentially decaying waveforms. The result is an exponential response to a volume change made by a user, where the change feels more pleasant and natural than a conventional linear response.

Abstract: An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

Abstract: An audio system (100) is provided with improved adaptive filter (206) to automatically adjust signal gain depending on the ambient noise level. The original music signal passes through a normalized adaptive filter (206), and is subtracted from the ambient room signal detected by a microphone (120), resulting in an error signal that is an estimate of the ambient noise. The error signal is used to update a set of adaptation coefficients so that the normalized adaptive filter more accurately simulates the room transfer function, resulting in an better estimate of the ambient noise. The audio system (100) is calibrated automatically upon initial use to determine adaptation coefficients and noise threshold level to prevent runaway gain. System parameters are adjusted using a controller (124) with a user-friendly interface (400).

Abstract: The invention relates to a device for the noise-dependent adjustment of the sound volume of a loudspeaker (8) fed with an electrical useful signal (2, 9) from a signal source (1) on a location of listening that is filled by noise (17). The inventive device comprises two sound volume adjusters (3, 5) that are connected in series and between the signal source (1) and the loudspeaker (8). One of said adjusters (3) can be manually controlled and the other (5) is controlled by a control signal (18). A noise detector (6) determines the noise level (17) and generates the control signal (18) for the other sound volume adjuster (5) in accordance with the noise level (17).

Abstract: Apparatus and a concommitant method for generating an approximation of a logarithmically attenuated value of a digital signal in predefined increments. Specifically, the apparatus comprises a shift register, a latch and an adder. The shift register selectively shifts the digital signal a first number of bits to the right to produce a first shifted signal and then a second number of bits to the right to produce a second shifted signal. The latch is connected to the shift register and latches the first shifted digital signal while the digital signal is being shifted to form the second shifted signal. The adder is connected to the shift register and the latch and adds the first shifted signal to the second shifted signal to produce the attenuated value of the input digital signal.

Abstract: A feedback controller, embedded in an FM audio processor, that controls the “integrated multiplex power” to the requirements of ITU-R 412 (2.51) is disclosed. This regulation specifies the maximum power, which is the same as the power produced by a sinewave modulating the carrier ±19 kHz (where ±75 kHz is 100% peak modulation). The controller applies the square of the multiplex signal to an integrator. A constant is removed from the integrator; representing the maximum power threshold. The output of the integrator is sampled periodically with a sample-and-hold circuit. A second integrator receives the output of the sample-and-hold. The output of the second integrator circuit is added to the threshold setting of a compressor, which determines the average power output of the compressor. The output of the compressor is applied through a peak controller to the input of the multiplex coder, closing a feedback loop, which controls the integrated multiplex power to a preset threshold.

Abstract: A feedback controller, embedded in an FM audio processor, that controls the “integrated multiplex power” to the requirements of ITU-R 412 (2.51) is disclosed. This regulation specifies the maximum power, which is the same as the power produced by a sinewave modulating the carrier ±19 kHz (where ±75 kHz is 100% peak modulation).

Abstract: A companding amplifier wherein compression data is provided to an expansion circuit along with audio data. An analog audio input signal is compressed and converted into digital data by a compression circuit. The compression circuit includes a data framer that produces a digital data stream that contains both the digital audio data and digital compression data. The resultant digital data stream can then be recorded on a medium such as a CD or DAT, or transmitted over a medium such as wire, fiber optic or radio waves. The digital data stream can then be read back or otherwise received by an expansion circuit that includes a data separator that separates the digital audio data and digital compression data. The digital audio data is then converted back to an analog audio signal, and the digital compression data is converted to an analog control signal that is used as the control signal for expansion.

Abstract: Level detectors detect levels of digital audio signals of individual channels. A controller predicts whether or not an overflow will take place corresponding to detected signals. When the controller predicts an occurrence of an overflow, attenuators attenuate a signal level of at least one channel. In addition, when variable length delaying devices vary the phases of signals, an overflow can be prevented.

Abstract: A handheld remote controller for a televisor that automatically prefixes a single-digit program channel selection, typically between 2 and 9, with a leading zero in response to a prolonged single digit keybutton actuation exceeding about 3 seconds or by an intended ersatz key entry, such as a star “*” key, either preceding or following the channel digit key entry. A HUSH control provides audio volume reduction intermediate between mute and normal volume levels, obtainable by pressing a HUSH keybutton, or automatically hushed concurrent with a program channel change. Maintaining a televisor's necessitous channel 3 or 4 selection when used in conjunction with a cable box, VCR or other intermediate device is assured by concurrently sending a reiterative channel 3 (or channel 4) tuning command to the televisor whenever a program channel change command or other function signal is dispatched to the cable box, VCR or other television signal tuning apparatus.

Abstract: A microphone output limiter which detects the amplitude of a microphone and produces a control signal which lowers the microphone's impedance when the amplitude of the audio signal exceeds a preselected threshold level. The output signal is detected at the output of the microphone or it may be detected at the output of a later pre-amplifier stage. Once a control signal is generated it is coupled to a variable impedance device which produces a shunt load impedance across the output of the microphone thereby lowering its impedance. The microphone, as well as the variable impedance device, may be coupled through impedance converters in order to facilitate component matching.

Abstract: A method and apparatus for combining analog and digital processing techniques to thereby capture a much wider dynamic range of an original input signal than an A/D converter can by itself, to thereby preserve a maximum amount of information from the original input signal, wherein a processing amplifier combines a level-dependent, soft logarithmic limiting function with a high-frequency pre-emphasis filter, such that the pre-emphasis helps to compensate for a disproportionate reduction of high frequency information when portions of the signal are in an overload region, and wherein digital de-emphasis of the digitized signal enables recovery of the dynamic range that was sacrificed to create the overload region.

Abstract: A method and arrangement for automatically limiting distortion in audio devices utilizes a distortion detector for determining a harmonic distortion factor and the output of the distortion detector is fed back to a gain regulation input of the audio device. The distortion detector determines the harmonic distortion factor of the output signal of the audio device in a frequency-dependent manner and, when the harmonic distortion factor exceeds a frequency-dependent, adjustable threshold value, the detector generates a control signal that reduces the gain of the audio device.