Patent classifications
H03G5/02
Method of operating an electronic device with a master fader and a plurality of slave faders
An electronic device with faders. In order to enable a user to readily ascertain which of a plurality of faders belongs to the same group and which one of the faders in the group is a master fader, indicators of all of the faders in the group are lighted when the thus-set master fader in the grouped faders is operated. When the user operates slave faders, the operation is revoked, and only the indicator corresponding to the master fader is lighted or blinked.
Equalization contouring by a control curve
A method for equalization contouring provides a reduction of equalization in certain frequency regions either by user control or by automated selection of frequency, without introducing artifacts. A control curve smoothly scales the magnitude of the equalization in the areas where less equalization is desired to obtain a contoured equalization. The control curve varies by frequency and may be defined specifically for every sampled frequency value of the equalization, may be a continuous function of frequency, or may be a function of control points at a select number of frequency points. The control curve may also have automatic inputs, e.g. a machine-detected cutoff frequency of a speaker may be used to determine a control point in the control curve. As another example, the reverberation time (e.g. RT60) may be used to determine a control point in the control curve. The result is a contoured equalization curve without sudden steps.
Methods and apparatus for improving understandability of audio corresponding to dictation
According to some aspects, a method for improving understandability of audio corresponding to dictation to assist a transcriptionist in transcribing the dictation is provided. The method comprises presenting a user interface to the transcriptionist, the user interface including at least one control that can be selectively set to one of a plurality of settings, receiving a selection of one of the plurality of settings via the at least one control, and compressing a dynamic range of at least a portion of the audio using at least one parameter value associated with the selected setting.
Mobile communication device capable of setting tone color and method of setting tone color
A mobile communication device and a method of setting tone color, which allow a user to set the tone color of received sound. Provided are a normal mode, which sets the equalizer using GCF standards stored in an internal memory or equalizer setting values selected by a provider, a country-specific mode, which uses country-specific setting, and a user mode, in which a user can set frequency-specific gains of the received sound, and one mode is selected from the provided mode, so that the tone color of the received sound can be adjusted according to the selection. Telephone speech quality can be optimized for user preference, network environments and language characteristics.
Electronic signal processor
An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.
Electrolarynx control button arrangement
An electrolarynx includes tone-producing circuitry, a power switch to turn on the circuitry, a control button (i.e., a pushbutton) to actuate the power switch, a pressure-sensitive-resistor (PSR) that is physically coupled to the pushbutton, and a mode switch. PSR resistance is dependent on the amount of pressure applied to the pushbutton, and the tone-producing circuitry is configured to respond to such variations in PSR resistance according to a user-selected mode of electrolarynx operation set by operation of the mode switch. Said modes preferably include multiple frequency-varying modes (FVMs) in which the frequency of the electrolarynx tone is varied with different sensitivities to variations in PSR resistance, and multiple volume-varying modes (VVMs) having different sensitivities. A preferred embodiment also includes a communications-link mode, for receiving control information from an external device, and a disabled mode. Preferably, the tone-producing circuitry includes a microcontroller component that is configured for electrolarynx operation under program control.
Filter coefficient group computation device and filter coefficient group computation method
A filter coefficient group computation device is configured from a means for performing inverse Fourier transform on a frequency characteristic inputted through an input means; a means for performing short-term Fourier transform on a numerical string obtained by the inverse Fourier transform; a means for performing windowing on a frequency domain signal, obtained by the short-term Fourier transform, using a function of which a window length shortens as frequency increases; a means for performing short-term inverse Fourier transform on the frequency domain signal after the windowing; a means for performing overlap addition on a numerical string obtained by the short-term inverse Fourier transform; and a means for determining a numerical string after the overlap addition as a filter coefficient group which forms a filter having the frequency characteristic inputted through the input means.
Filter coefficient group computation device and filter coefficient group computation method
A filter coefficient group computation device is configured from a means for performing inverse Fourier transform on a frequency characteristic inputted through an input means; a means for performing short-term Fourier transform on a numerical string obtained by the inverse Fourier transform; a means for performing windowing on a frequency domain signal, obtained by the short-term Fourier transform, using a function of which a window length shortens as frequency increases; a means for performing short-term inverse Fourier transform on the frequency domain signal after the windowing; a means for performing overlap addition on a numerical string obtained by the short-term inverse Fourier transform; and a means for determining a numerical string after the overlap addition as a filter coefficient group which forms a filter having the frequency characteristic inputted through the input means.
ALTERING AUDIO TO IMPROVE AUTOMATIC SPEECH RECOGNITION
Techniques for altering audio being output by a voice-controlled device, or another device, to enable more accurate automatic speech recognition (ASR) by the voice-controlled device. For instance, a voice-controlled device may output audio within an environment using a speaker of the device. While outputting the audio, a microphone of the device may capture sound within the environment and may generate an audio signal based on the captured sound. The device may then analyze the audio signal to identify speech of a user within the signal, with the speech indicating that the user is going to provide a subsequent command to the device. Thereafter, the device may alter the output of the audio (e.g., attenuate the audio, pause the audio, switch from stereo to mono, etc.) to facilitate speech recognition of the user's subsequent command.
ALTERING AUDIO TO IMPROVE AUTOMATIC SPEECH RECOGNITION
Techniques for altering audio being output by a voice-controlled device, or another device, to enable more accurate automatic speech recognition (ASR) by the voice-controlled device. For instance, a voice-controlled device may output audio within an environment using a speaker of the device. While outputting the audio, a microphone of the device may capture sound within the environment and may generate an audio signal based on the captured sound. The device may then analyze the audio signal to identify speech of a user within the signal, with the speech indicating that the user is going to provide a subsequent command to the device. Thereafter, the device may alter the output of the audio (e.g., attenuate the audio, pause the audio, switch from stereo to mono, etc.) to facilitate speech recognition of the user's subsequent command.