Abstract: A band-splitting time compensation signal processing device 2 includes a band-splitting circuit 211 for extracting, after extracting a signal of a high-frequency band component or low-frequency band component from an input signal, a signal of the low-frequency band component or high-frequency band component by subtracting the signal of the high-frequency band component or low-frequency band component from the input signal; a delay circuit 212 for delaying, for adjusting arrival time, at least one of the high-frequency band component and low-frequency band component output from the band-splitting circuit 211; and a mixing circuit 213 for combining the high-frequency band component or low-frequency band component output from the delay circuit 212 with the low-frequency band component or high-frequency band component output from the band-splitting circuit 211.

Abstract: A frequency equalization system is provided that substantially equalizes the room frequency responses generated by at least one loudspeaker within a listening area so that the frequency responses in the listening area are substantially constant and flat within a desired frequency range with minimum signal latency. The frequency equalization system may use multiple microphones to measure the audio signals of one or more subwoofers to achieve an improved bass response that is flat across the relevant frequency range. The system employs an algorithm that is a closed-form, non-iterative, mathematical solution and features short computation time with reduced delays.

Abstract: The present disclosure provides a digital audio signal processing system that comprises a set of delay lines, allpass and lowpass filters to achieve the reverberation effect. The present disclosure further provides a method for generating and controlling digital reverberations for audio signals. The reverberation generated will have an increasing echo density in the time domain and a faster decay of high frequency signals than low frequency signals. The controlling mechanism of reverberation generation is realized through the extraction of the real environment characteristics.

Abstract: Methods are provided for solid-state implementation of a vacuum tube replacement device (VTRD) that includes a communication interface. The communication interface enables the VTRD to command or retrieve information regarding VTRD performance, function, and status. A VTRD may communicate with other VTRDs in a target amplifier retrofitted with compatible VTRDs and to an external user interface for programming or query/response. The methods presented for the vacuum tube replacement device system(s) are equally useful for non-vacuum tube systems such as audio amplifier circuits.

Abstract: The invention relates to adaptive directional systems, and more particularly to a method and apparatus for producing adaptive directional signals. The invention may be applied to the provision of audio frequency adaptive directional microphone systems for devices such as hearing aids and mobile telephones. The method involves constructing the adaptive directional signal (46) from a weighted sum of a first signal (42A) having an omni-directional polar pattern and a second signal (42B) having a bi-directional polar pattern, wherein the weights are calculated to give the combined signal a constant gain in a predetermined direction and to minimize the power of the combined signal. The method has particular application in producing signals in digital hearing aids, the predetermined direction being in the forward direction with respect to the wearer.

Abstract: The invention relates to sonic steerable antennae and their use to achieve a variety of effects. The invention comprises a method and apparatus for taking an input signal, replicating it a number of times and modifying each of the replicas before routing them to respective output transducers such that a desired sound field is created. This sound field may comprise a directed beam, focus beam or a simulated origin. Further, “anti-sound” may be directed so as to create nulls (quiet spots) in an already existing sound field. The input signal replicas may also be modified in way which changes their amplitude or they may be filtered to provide the desired delaying.

Abstract: A vehicle engine harmonic enhancement system. A method for operating the system includes providing a first engine harmonic enhancement audio signal and equalizing the first engine harmonic enhancement audio signal separately for each of the plurality of loudspeakers to provide individually equalized loudspeaker engine harmonic enhancement audio signals. A vehicle audio system includes a plurality of loudspeakers, an entertainment audio system, and an engine harmonic enhancement system. A method includes equalizing an entertainment audio signal to provide an equalized entertainment audio signal; providing an engine harmonic enhancement audio signal; equalizing the engine harmonic enhancement audio signal separately from the entertainment audio signal to provide an equalized engine harmonic enhancement audio signal; and combining the equalized entertainment audio signal with the equalized engine harmonic enhancement audio signal.

Abstract: A user interface device provides secure access to equipment. An audio reception device is adapted to receive an audio input comprising background noise and to receive an audio user code from a user. A user input device receives a manual input comprising a manual user code from the user. A control unit stores one or more target user codes and receives an audio input and determines a noise level of the background noise. An instruction message is generated to inform the user to enter the manual user code in response to the noise level of the background noise exceeding a predetermined threshold level. An output device provides the instruction message to the user. A control unit provides secure access to the equipment in response to at least one of the audio user code and the manual user code.

Abstract: According to one embodiment, a sound corrector includes a signal outputter, a response signal, a frequency specifier, a coefficient specifier, a filter, and an outputter. The signal outputter outputs a measurement signal to measure acoustical properties of an object to be measured. The response signal receiver receives a response signal from the object in response to the measurement signal. The frequency specifier specifies a resonant frequency at a resonance peak from the response signal. The coefficient specifier specifies a correction coefficient of a correction filter for reducing the resonant frequency based on the specified resonant frequency. The filter performs filtering on a signal to be output to the object using the correction filter with the correction coefficient. The outputter outputs the signal having undergone the filtering to the object.

Abstract: A method and system for active noise cancellation is provided. The system employs subband processing, and preferably implements over-sampled filterbank. The system is applicable to adaptive noise cancellation, adaptive echo cancellation for portable listening devices, such as headsets and other similar listening devices.

Abstract: A noise reduction system and a noise reduction method are provided. The noise reduction system comprises a uni-directional microphone, an omni-directional microphone and a signal processing module. The signal processing module comprises an adaptive noise control (ANC) unit, a main noise reduction unit and an optimizing unit. The uni-directional microphone senses a first audio source to output a first audio signal, and the omni-directional microphone senses a second audio source to output a second audio signal. The ANC unit executes an adaptive noise control to output an estimated signal according to the first audio signal and the second audio signal. The main noise reduction unit executes a main noise reduction process to output a de-noise speech signal according to the estimated signal and the second audio signal. The optimizing unit executes an optimizing process to output an optimized speech signal according to the de-noise speech signal.

Abstract: According to an aspect of the invention, there is provided a noise suppressing apparatus comprising: a fifth unit configured to calculate a gain for noise suppression, based on the first signal-to-noise ratio for each frequency band and the second signal-to-noise ratio for an entire frequency band; an eighth unit configured to calculate an upper limit value of a noise suppression amount for each frequency band, based on the second signal-to-noise ratio; a ninth unit configured to calculate the noise suppression amount for each frequency band, based on the first signal-to-noise ratio; and a tenth unit configured to limit, based on the upper limit value, the noise suppression amount so as to calculate the gain.

Abstract: A device for acoustic analysis has a first hearing device with a first sound input and a first sound output and a second hearing device with a second sound input and a second sound output. The first hearing device is in acoustic communication with the second hearing device. The first hearing device can analyze the acoustic communication and output a corresponding result. Consequently, two hearing aids, for example, can test each other, and permit a user to check hearing devices and, in particular, hearing aids in a simple fashion without the hearing aid wearer having to visit an audiologist for the test.

Abstract: A signal processing system includes a level dependent bass management system. The level dependent bass management system utilizes audio input signal level information to apply at least one of multiple, available bass management solutions to generate one or more output signals from the audio input signal. In at least one embodiment, initially the level dependent bass management system boosts components of the audio input signal in the low frequency range by an amount sufficient to at least partially compensate for low frequency attenuation of the first speaker without exceeding one or more acceptable limitations of the signal processing system. If boosting alone cannot completely compensate for low frequency attenuation of the first speaker without exceeding one or more acceptable limitations of the signal processing system, the level dependent bass management system processes the audio input signal using an alternate low frequency management solution.

Abstract: A sound element is generated by synthesizing, from a base sound, a frequency component of a sinusoidal wave one octave higher than the base sound. The base sound is a frequency component of one sinusoidal wave with an integer multiple of periods thereof matching a sample count represented by a power of 2. Sound elements having a frequency serving as a musical scale in a temperament are selected from the sound elements. The selected sound element is outputted in a predetermined pattern of time and a musical scale, so that a test tone is produced in a melody-like fashion.

Abstract: In accordance with one aspect of the present invention, a method for detecting a change in the performance of an audio signal processing path is disclosed. The method comprises: selecting a characteristic of a received audio signal indicative of its energy content; determining first and second predetermined values of the selected energy characteristic at respective first and second audio signal frequency bands; calculating a ratio of the first and second predetermined values for a reference time period and a test time period; and comparing the ratio at the reference time period with the ratio of the test time period to determine a performance change in the audio path.

Abstract: There is described a hearing device especially for musicians. For this purpose a hearing device is proposed, in whose housing a music device is integrated with which tones or signals can be emitted via the signal processor and the loudspeaker to the hearing device wearer. This music device can, for example, be a tuner with tone pitch analysis or sound playback, a metronome and/or a tone pitch processing unit. A remote control for receiving pieces of music and for playback via a hearing device may also be provided.

Abstract: Methods and apparatus to provide an equalizer for analog adaptive control are disclosed. An example equalizer described herein includes a high frequency amplifier to receive an input signal and to amplify a high frequency portion of the input signal, a low frequency amplifier to receive the input signal and to amplify a low frequency portion of the input signal, and a weight factor controller to control a gain of the high frequency amplifier and a gain of the low frequency amplifier.

Abstract: Four speakers are matrix-connected to each other and a conversion circuit for converting audio signals of four channels into pulse width modulation signals of four channels that are time-divided so as to be complementary to matrix connection are provided. Four class-D output amplifiers receive the corresponding pulse width modulation signals output from the conversion circuit and outputs from the output amplifiers are supplied to the speakers.

Abstract: In an electronic apparatus, microphones are arranged parallel to one side of a display screen so as to be spaced apart from each other on the front surface of a display whose display screen is exposed. Together with the microphone, a microphone is arranged parallel to other side different from one side of the display screen. Of the microphones, the microphones are selected when one side of the display screen serves as a bottom, and the microphones are selected when other side of the display screen serves as the bottom. A voice input process is then executed.