171results 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.

Amplitude and phase error correction in a transceiver circuit is provided. In embodiments disclosed herein, a transceiver circuit is configured to equalize an input vector to thereby correct amplitude-amplitude (AM-AM) and amplitude-phase (AM-PM) errors across a modulation bandwidth of the wireless communication circuit. Unlike conventional methods where complicated memory digital predistortion (mDPD) coefficients must be defined and calibrated for each modulation frequency within the modulation bandwidth, the transceiver circuit is configured herein to eliminate modulation frequency dependency of the AM-AM and AM-PM errors. As a result, it is possible to correct the AM-AM and AM-PM errors across the modulation bandwidth with reduced complexity to thereby improve efficiency and linearity of the wireless communication circuit.

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.

An audio system, including at least one approximating circuit, the at least one approximating circuit configured to apply an approximated gain to the input signal, the least one approximating circuit outputting an approximated signal; a limiter configured to reduce a magnitude of the input signal by a gain determined by an amount the magnitude of the approximated filtered signal exceeds a predetermined threshold, the limiter outputting a limited input signal; a filter being respectively configured to receive the limited input signal and to respectively apply a filter gain, the filters outputting a plurality of filtered signals, wherein the approximated gain is substantially similar to the filter gain of one of the plurality of equalization filters; and a speaker array comprising a plurality of speakers, each speaker configured to receive a respective filtered signal of the plurality of filtered signals and to transduce the filtered signal to an audio signal.

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 transmission line includes an equalization circuit. The equalization circuit is a second-order equalization circuit having a first loop at a gain element and a second loop at the gain element. The first loop may include a first compensation capacitor, and the second loop may include a second compensation capacitor and a resistor. The second order equalization circuit may allow for improved performance with respect to gain as well as reduced power usage.

Techniques for shaping sound based on a speaker orientation are disclosed. In an embodiment, an audio data stream is obtained by a zone player having one or more speaker drivers, an orientation of the zone player is determined, and sound is reproduced by the zone player based on the orientation. The sound may be further shaped based on other states of the zone player in addition to orientation. The overall sound may be shaped from one zone player or from a collection of zone players.

Some noise compensation methods involve: receiving a microphone signal corresponding to ambient noise from a noise source location in or near an audio environment; determining or estimating a listener position in the audio environment; and estimating at least one critical distance, the critical distance being a distance from the noise source location at which the directly propagated sound pressure equals the diffuse-field sound pressure. Some examples involve estimating whether the listener position is within the at least one critical distance, and performing the noise compensation method for the ambient noise based at least in part on the estimation of whether the listener position is within the critical distance.

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.

An apparatus (100) for processing an input audio signal (122) for reproduction by a sound transducer (130) comprises an equalization parameter determinator (110) for determining a set of equalization parameters (112) and an equalizer (120) configured to equalize the input audio signal (122), to obtain an equalized audio signal (124). Different concepts for the determination of the set of equalization parameters (112) comprise an image recognition, an evaluation of an identification signal which is provided by the sound transducer via an audio connection, and a measurement of the impedance of the sound transducer over frequency. Also, an upload functionality and a download functionality are provided.

A decision feedback equalizer and a method for setting a decision feedback equalizer, the decision feedback equalizer comprising a feedforward equalizer, a feedback equalizer, a decision device, and a decision adjustment device. The feedforward equalizer is configured to generate a feedforward output signal based on an input signal. The feedback equalizer is coupled to the feedforward equalizer and configured to generate a feedback output signal based on a decision output signal. The decision device is coupled to the feedforward equalizer and the feedback equalizer and is controlled by a decision adjustment parameter, wherein the decision device is configured to make a decision based on a sum of the feedforward output signal and the feedback output signal to generate the decision output signal. The decision adjustment device is coupled to the decision device and is configured to adjust the decision adjustment parameter based on a system sleep state.

A system for determining an error in an electrical signal is presented, including: an input configured to receive an input signal; an output configured to provide an output signal; an equalizer coupled to the input; an amplifier coupled to the equalizer; and an error determination circuit coupled to the output, the input, and the equalizer.

The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

A method of volume control is disclosed, comprising: playing audio through a speaker of a headset at a volume setting; receiving a microphone signal that includes noise of an acoustic environment capt

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.