Abstract: A method and system is described to improve the reproduction of sound of an acoustic musical instrument. According to one embodiment, a first microphone is placed at a proximate location to the musical instrument to pick up the sound of the musical instrument. The sound as picked up by the first microphone is compared to a reference sound of the instrument (e.g., the sound of the instrument as perceived at a normal listening position). Based on this comparison, a tailor-made equalizer is designed to compensate for the differences between the sounds as picked up by the first microphone and the reference sounds of the musical instrument. Accordingly, using the tailor-made equalizer allows the reproduction of sound from the first microphone to have a quality similar to that of the reference sound of the musical instrument. In an implementation of the above system, a filter arrangement is provided having a low-pass and a high-pass filter that allows separate control of the frequency and/or gain for each filter.

Abstract: Manually-operable and automatically-driveable fader operators are provided in corresponding relation to a plurality of channels, and a plurality of the fader operators are grouped in such a manner that the fader operators in a same group are operable collectively in an interlocked fashion. On a display screen, a window of a given group is opened in response to selection operation by a user, and a graphic image indicative of settings of the individual fader operators belonging to the given group is displayed. In response to operation by the user on the image displayed on the window of the given group and indicative of the settings of the individual fader operators, an image for effecting a change in the settings is displayed, and the individual fader operators belonging to the given group are automatically driven, to thereby change the settings. Such arrangements allow a grouping state of the operators to be identified visually with an increased ease and also facilitate setting operation.

Abstract: Improved audio source separation is provided by providing an audio dictionary for each source to be separated. Thus the invention can be regarded as providing “partially blind” source separation as opposed to the more commonly considered “blind” source separation problem, where no prior information about the sources is given. The audio dictionaries are probabilistic source models, and can be derived from training data from the sources to be separated, or from similar sources. Thus a library of audio dictionaries can be developed to aid in source separation. An unmixing and deconvolutive transformation can be inferred by maximum likelihood (ML) given the received signals and the selected audio dictionaries as input to the ML calculation. Optionally, frequency-domain filtering of the separated signal estimates can be performed prior to reconstructing the time-domain separated signal estimates. Such filtering can be regarded as providing an “audio skin” for a recovered signal.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

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: It is intended to improve and automate the calculation of calibration filters connected downstream from the microphones of an RGSC beamformer. To this end it is proposed that an adaptive calibration filter calculation unit be used, by means of which calibration filters are calculated from the output signals of adaptive blocking filters such that the power of an output signal of a blocking filter subtracted from a reference signal and filtered by means of a calibration filter respectively is minimized. The calibration filters connected downstream from the microphones are then replaced by the calibration filters thus determined.

Abstract: A microphone circuit includes an amplifier for amplifying an electric signal output from an electroacoustic transducing module, an external power supply circuit capable of supplying power for driving at least the amplifier from the outside, a light source that is turned on when an external power is supplied through the external power supply circuit, and a photo relay having contacts that are turned on and off in response to blinking of the light source. The contacts of the photo relay are connected so as to output an output signal of the amplifier as a microphone output when the light source is turned on and output an output signal of the electroacoustic transducing module as the microphone output when the light source is turned off.

Abstract: A handsfree communication system includes microphones, a beamformer, and filters. The microphones are spaced apart and are capable of receiving acoustic signals. The beamformer compensates for propagation delays between the direct and reflected acoustic signals. The filters are configured to a predetermined susceptibility level. The filter process the output of the beamformer to enhance the quality of the received signals.

Abstract: A two way communication system that does not require the use of a device by one of the communicators. The system employs an audio spotlight to project a signal to a specific location, where one person is, and a laser audio receiver to receive the response. Both the audio spotlight and the laser audio receiver are directed by a user seeking to communicate with the person at long range.

Abstract: A system and a methods for correcting, simultaneously at multiple-listener positions, distortions introduced by the acoustical characteristics includes warping room responses, intelligently weighing the warped room acoustical responses to form a weighted response, a low order spectral fitting to the weighted response, forming a warped filter from the low order spectral fit, and unwarping the warped filter to form the room acoustical correction filter.

Abstract: An apparatus having a microphone and a mute circuit (not shown in detail here) operable to be actuated by a rotating door assembly. The apparatus provides a mute circuit suitable to mute the audio microphone when the rotating door assembly is in a closed position and unmute the audio microphone when the rotating door assembly is not in the closed position. Additionally, the mute circuit is impervious to generating audible disturbances in the microphone signal when the rotating assembly door is rotated between open and closed positions. This advantageously allows for the microphone to essentially transition to off when the rotating door assembly is closed and transition to on when the rotating door assembly is open and do so without additional audio disturbance.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: A speaker array, includes a plurality of speakers which are linearly arranged at a predetermined interval; and one-dimensional digital filters which are provided to correspond to the speakers respectively, in which predetermined filter coefficients are set previously, and which apply a filtering process to input sound data in response to the filter coefficients to output. Sound data derived by applying a digital conversion to input sound signals are supplied to respective one-dimensional digital filters. Sound signals derived by applying an analog conversion to the sound data output from respective one-dimensional digital filters are supplied to corresponding speakers to output a sound in response to the sound signals.

Abstract: Headsets (20, 50, 60, 71) provide surround sound and full 3 dimensional effects to a user to simulate the effects of direction and sound source. Advantages include: surround sound effect without the limitations of Head-Related Transfer Functions, and pinna effect customized to each user's ears; horn shape tube outlets (32) to create an efficient transmission of sound; ability to space the headset speakers (22, 24) away from the user's ear and to maintain sound quality by addition of a chamber (28) behind the speakers (22, 24), with the aid of tube sound guides (23, 25) and the horn shaped outlets; and, no need for electronic hardware to process the electrical signals to create desired effects as placement of the speakers creates the correct timing, and damping material (34) in the tubes between speakers creates a desired intensity drop. Head tracking capability is also provided.

Abstract: Apparatus and a corresponding method for processing speech signals in a noisy reverberant environment, such as an automobile. An array of microphones (10) receives speech signals from a relatively fixed source (12) and noise signals from multiple sources (32) reverberated over multiple paths. One of the microphones is designated a reference microphone and the processing system includes adaptive frequency impulse response (FIR) filters (24) enabled by speech detection circuitry (21) and coupled to the other microphones to align their output signals with the reference microphone output signal. The filtered signals are then combined in a summation circuit (18). Signal components derived from the speech signal combine coherently in the summation circuit, while noise signal components combine incoherently, resulting in composite output signal with an improved signal-to-noise ratio. The composite output signal is further processed in a speech conditioning circuit (20) to reduce the effects of reverberation.

Abstract: The invention relates to a method for assessing interfering noise in motor vehicles, according to which noise occurring during a predefined measuring time is divided into different frequency ranges, the changes in level relative to the background noise are determined within said frequency ranges, and the determined changes in level are evaluated.

Abstract: A method for processing the signals from two or more microphones in a listening device, and a listening device for conducting the method which has a casing holding the microphones, a signal processing unit which provides an output signal corresponding to the microphone signals and suited to the user's hearing, and a receiver unit for delivering the output signal to the user whereby the signals from the microphones are analyzed in order to detect when the casing of the listening device is being touched, whereby further the signal processing of the signal processing unit changes whenever touching of the casing is detected.

Abstract: A microphone apparatus for processing and outputting an output signal of a microphone array including at least nine microphones includes a directivity function processing circuit that converts the output signal of the microphone array into a unidirectional signal and that outputs the unidirectional signal. The directivity function processing circuit expands a directivity function whose variable is an incident angle of an acoustic wave into a Fourier series up to at least third order. The variable in the expanded expression is produced from output signals of the microphones forming the microphone array.

Abstract: A method is provided for computing frequency responses of crossover filters for multi-way loudspeakers. The method prescribes driver coordinates for drivers in the multi-way loudspeaker, prescribes an attenuation function for the sound pressure level at a desired angle, computes the crossover frequencies using a point source model and computes the frequency responses in intervals defined by the crossover frequencies.

Abstract: A signal generating module for an electronic apparatus is provided. The signal generating module includes a processor, a roller, a resistant film, a conducting member, a first lead and a second lead. The roller has a rotary center and is rotably connected to the electronic apparatus thereby. The conducting member is connected to the rotary center to rotate together with the roller. The conducting member has an end including a conducting contact point. The conducting contact point is configured to rotate around the rotary center along a motion path accompanying the rotation of the roller, wherein the motion path is disposed at a first area of the resistant film. The first lead is coupled to the processor and in contact with the resistant film. The second lead is coupled to the processor, and coupled to the conducting member via the rotary center.