105results about "Automatic tone/bandwidth control" patented technology

Embodiments are provided for receiving a request to output audio at a first speaker and a second speaker of an electronic device, determining that the electronic device is oriented in a portrait orientation or a landscape orientation, identifying, based on the determined orientation, a first equalization setting for the first speaker and a second equalization setting for the second speaker, providing, for output at the first speaker, a first audio signal with the first equalization setting, and providing, for output at the second speaker, a second audio signal with the second equalization setting.

Techniques are described for wirelessly receiving, at first and second speaker assemblies, signals representing left and right audio channels from each of first and second sources, and using the signals, simultaneously playing, on both speaker assemblies, music, computer game audio, computer chat audio, and speech picked up by speaker microphones. User-adjusted and non-user-adjusted EQ may be implemented on various portions of the signal processing.

The invention relates to a method for calibrating a loudspeaker system in an acoustic environment. The method comprises steps of providing a local frequency response based on a recorded test signal and providing a target frequency response. A difference frequency response is established based on a difference between the local frequency response and the target frequency response. A list of exclusion frequency ranges associated with minima of the local frequency response is generated and one or more filter frequency ranges which are non-overlapping with the exclusion frequency ranges are identified. A target filter frequency is selected from the identified filter frequency ranges and a calibration filter related to the target filter frequency is implemented in an equalizer of the loudspeaker system. The invention further relates to a loudspeaker system arranged to carry out the above method for calibrating a loudspeaker system.

A computer system for manipulating, combining, or composing dynamic sounds accesses a package of one or more music stems. The computer system then receive an input variable from one or more vehicle sensors. The one or more vehicle sensors measure an aspect of driving parameters of a vehicle. In response to the input variable, the computer system generate a particular audio effect with the one or more music stems.

In various embodiments, the disclosed techniques automatically change the audio equalization levels of audio content. When an audio content is requested for playback, the equalization manager of an audio system extracts the frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines sound pressure level (SPL) metrics of each frequency component. The equalization manager generates a custom equalization setting using gains calculated for each frequency band based on the SPL metrics. The equalization manager then adjusts the audio segment by applying the custom equalization setting to generate a modified segment (e.g., applying the calculated gains to each of the frequency bands). The modified segment is then output as audio to a user. The process then repeats for additional segments of the audio.

In various embodiments, the disclosed technology automatically alters audio equalization levels of audio content. When a request to play audio content is made, an equalization manager of an audio system extracts frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines a sound pressure level (SPL) indicator for each frequency component. The equalization manager generates a customized equalization setting based on the SPL indicators using gains computed for each frequency band. The equalization manager then adjusts the audio segment by applying the customized equalization setting to generate a modified segment (e.g., applying the computed gains to each of the frequency bands). The modified audio segment is then output as audio to a user. The process is then repeated for additional segments of the audio.

A method and corresponding system for generating harmonics based on an input signal are provided. The method includes (S1) obtaining one or more frequency band signals from the input signal; generating (S2) a first amplitude scaling signal comprising one or more harmonic components based on at least one or each of the frequency band signals, the amplitude of the first amplitude scaling signal being proportional to the amplitude of the input signal; generating (S3) a normalized signal based on at least one or each of the frequency band signals, the amplitude of the normalized signal being independent of the amplitude of the input signal; and for at least one or each of the frequency band signals, multiplying the normalized signal by the corresponding first amplitude scaling signal to generate a second amplitude scaling signal comprising one or more harmonic components; and for at least one or each of the frequency band signals, generating (S4) an output signal based on the first amplitude scaling signal comprising one or more harmonic components and the second amplitude scaling signal comprising one or more harmonic components.

A method of equalising a signal derived from an input device, the method comprising: receiving the signal; applying an order-statistic filter to the signal in the frequency domain to generate a statistically filtered signal; equalising the received signal based on the statistically filtered signal to generate an equalised signal.

To dynamically change settings of an equalizer in accordance with performance.SOLUTION: Provided is a signal processing device including: an acquisition unit that acquires fingering information concerning fingering of a stringed instrument; and a setting unit that sets, as parameters, a frequency and level adjustment information in an equalizer. The frequency is a frequency corresponding to the fingering information. The level adjustment information is information for adjusting a signal level of a sound signal of the frequency in accordance with the signal level. The setting unit changes the frequency when the fingering information is changed.SELECTED DRAWING: Figure 1

Examples disclosed herein relate to playback device calibration. A calibration state variable may represent a calibration state of a playback device. A control device may display an indications of the calibration state. For instance, in a first instance, the control device may display an indication that a first playback device is in an uncalibrated state. Further, in a second instance, the control device may display an indication that the first playback device is in an calibrated state, perhaps after instructing the first playback device to initiate calibration.

An arithmetic device executes a method for acoustic adjustment including: setting an initial value of a gain value of a preamplifier based on a difference between energies of the test sound source before and after correction by an equalizer to an input sound; amplifying a volume of the test sound source by the preamplifier using the initial value, calculating a loudness value and a clipping occurrence rate of the test sound source corrected by the equalizer, and determining whether calculation results of the loudness value and the clipping occurrence rate are each less than a predetermined threshold; and reducing the initial value of the gain value by a specified value in a case where the calculation result of the loudness value or the clipping occurrence rate is equal to or greater than the predetermined threshold.

Method in an electrical audio signaling circuit comprising a feedback loop (280) with a digital filter (285) coupled to a current-to-digital-to-analog converter (240, 320), the method comprising: providing (560, 625) an output signal from the current-to-digital-to-analog converter (240, 320) to analog elements of the electrical audio signaling circuit, wherein the output signal from the current-to-digital-to-analog converter (240, 320) is based on a reference signal input to the current-to-digital-to-analog converter (240, 320) while no output from the digital filter (285) is input to the current-to-digital-to-analog converter (240, 320); comparing (565, 630) an output signal of the electrical audio signaling circuit with a reference; and calibrating (570, 635) the electrical audio signal circuit, such that the output signal of the electrical audio signal circuit corresponds to the reference,wherein the calibration of the electrical audio signal circuit enables more precise control of a cutoff frequency of a microphone signal, while the output of the digital filter (285) is input into the current-to-digital-to-analog converter (240, 320).

Method for adapting an audio output to users of a vehicle, wherein the vehicle is designed with separate audio zones for different seating positions, in which - for different users of the vehicle, an individual set of parameters describing the frequency-dependent hearing ability of the respective user is determined (2); and - by means of the respective individual parameter set in the vehicle for multiple users, depending on the seating position, a suitable correction of the audio output signal is made by a frequency response and dynamic range adjustment for different frequency ranges (4).

A device for adjusting the volume of an audio signal in an audio system of a motor vehicle is proposed, wherein the audio system has an input for receiving the audio signal from an input source and an output for outputting the audio signal, wherein the volume of the audio signal is adjustable by means of a control of the audio system, wherein the device has a control unit configured to determine an audio signal gain based on the current volume of the audio signal and based on a current speed of the motor vehicle, to determine a first volume increment factor based on the set volume, and to adjust the volume of the audio signal using the audio signal gain and the first volume increment factor.The control unit is further designed to determine a low-frequency amplification based on the current speed of the vehicle, to determine a second volume increment factor based on the set volume, and to adjust the low-frequency volume of the audio signal using the low-frequency amplification and the second volume increment factor, and / or the control unit is designed to adjust an output of the audio signal via the vehicle's structure-borne sound transducers based on the current speed of the vehicle using a structure-borne sound transducer parameter.

The invention discloses a gain control method of an audio amplifier. The gain control method comprises the following steps: detecting the amplitude of an output signal of the audio amplifier; if the amplitude of the output signal is smaller than a first preset threshold value, detecting whether the output signal passes through a zero point or not; and after detecting that the output signal passes through the zero point, reducing the gain of the audio amplifier. The noise of the audio amplifier can be reduced.

The invention discloses an audio equalization method and device, a storage medium and a program product, and belongs to the technical field of audio. The method comprises the steps of obtaining a frequency spectrum of a target audio and a frequency spectrum of a reference audio, generating an EQ curve representing a frequency spectrum difference between the target audio and the reference audio, determining a statistical characteristic of the frequency spectrum difference, determining a compression parameter for performing range compression on the frequency spectrum difference based on the statistical characteristic, and performing range compression on the frequency spectrum difference based on the compression parameter and a soft knee compression algorithm. And performing optimization processing on the EQ curve to obtain an optimized EQ curve, and performing equalization processing on the audio signal of the target audio based on the optimized EQ curve to obtain an equalized audio signal of the target audio. Through soft knee compression processing, the spectrum distortion possibility of the balanced audio signal is reduced.

A system to provide dynamic equalization for power management. In some embodiments, the system comprises a powered communication port configured to receive audio data and line power, and one or more amplifiers configured to drive one or more speakers. The one or more amplifiers have a peak power consumption that exceeds the maximum power of the line power. The system also comprises power supply circuitry, including at least one capacitor, and an equalizer. The equalizer is configured to generate equalized audio data by: performing a first attenuation on a spectral region of the audio data based on an estimated power budget; applying a temporary second attenuation (e.g., a volume reduction) based on the capacitor voltage; applying a first gain to reverse a portion of the volume reduction over a first time period; and applying a second gain to reverse a remainder of the volume reduction over a second time period.

In various applications, the system provides a method for processing audio signals, including: receiving, by a processor, a digital audio signal from a recorded audio file; analyzing, by the processor, the digital audio signal to identify pitch distortion caused by changes in momentary sound level; determining, by the processor, an amount of compensation of the audio signal to correct the identified pitch distortion; dynamically adjusting, by the processor, the digital audio signal by the compensation amount to correct the identified pitch distortion; and outputting, by the processor, the digital audio signal to an audio transducer device of a listener to improve a listening experience for the listener of the recorded audio file.

A method for processing an audio signal is disclosed. According to an embodiment of the method, prototype filter magnitude response information is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are calculated for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the calculated gains are applied to the audio signal.

A signal processing system for adjusting an audio signal, the signal processing system configured to: receive an audio signal for adjustment at a signal input; at a function generator, receive a selection of a musical key and scale, the musical key and scale including musical notes, and generate an audio signal processing function, configured to adjust the amplitude of frequencies of an audio signal, the audio signal processing function being dependent on the selected musical key and scale; apply the generated audio signal processing function to the received audio signal to generate an adjusted audio signal; and, output the adjusted audio signal at a signal output.

Circuit arrangement (200), comprising: a sensor, set up to provide an analog signal (204); an analog-to-digital converter (202), set up to receive the analog signal (204) and provide an initial signal (206); a first filter (208) configured to receive a signal based on the first signal (206) and to provide a second signal (210); where the first filter (208) is set up such that: • that the second signal (210) is passed through unamplified in a frequency range from 20 Hz to 10 kHz, and • that the second signal (210) has a gain greater than 0 dB at least from a specified frequency greater than 20 kHz; and a second filter (214) connected to the analog-to-digital converter (202) and configured to provide a third signal (216) based on the first signal, wherein the second filter (214) is configured as a frequency-selective filter to process the signal received by the second filter, that the amplitude of the signal received by the second filter (214) is reduced in a specified frequency range, wherein the specified frequency range has a frequency greater than 20 kHz.

A sound capture system can include a sound capture device configured for placement within a sound-generating device. The sound capture system can include a suspension system configured to suspend the sound capture device within the sound-generating device. The suspension system acoustically decouples the sound capture device from the sound-generating device. The sound capture device includes a plurality of transducers configured to generate a plurality of audio signals concurrently from sound generated by the sound-generating device. In another aspect, the sound capture system includes a signal processing circuit.

In various embodiments, a sound modification application selectively modifies one or more sounds included in one or more audio signals. In operation, the sound modification application determines classifications associated with multiple sounds included in one or more audio signals. The sound modification application selects a first frequency sub-band of a first sound included in the multiple sounds based on a first classification associated with the first sound. The sound modification application then modifies the first frequency sub-band of the first sound, without modifying at least a second frequency sub-band of the first sound, to generate a modified audio signal.

Various examples relate to a frequency-compensated amplifier circuit including a first multi-stage amplifier including a first amplifier input node, a first amplifier output node, and a first amplifier intermediate node. A first feedback path between the first amplifier input node and the first amplifier output node includes a feedback resistor. A second feedback path between the first amplifier output node and the first amplifier intermediate node includes a first capacitor and a portion of the feedback resistor. A first switch circuit can be electrically coupled to the first capacitor and the feedback resistor. The first switch circuit can have a first state in which the first capacitor is coupled to a first tap point of the feedback resistor and the portion of the feedback resistor has a first value. The first switch circuit can also have a second state in which the first capacitor is coupled to a second tap point of the feedback resistor and the portion of the feedback resistor has a second value different from the first value.

A system is described that comprises a computing device comprising one or more processors and a memory. The computing device is configured to receive user input and to analyze the user input to generate control data. The computing device is further configured to analyze the audio input to generate signal data that are characteristic of the audio input and to analyze previous tuning data to generate reference data. The computing device is also configured to process at least the control data, the signal data, and reference data to generate tuning data. In addition, the computing device is configured to process audio input to generate audio output based on at least one audio model and the tuning data.