437results about "Low frequency amplifiers" patented technology

A circuit comprises a first half-bridge power stage including a first set of switches and a second half-bridge power stage including a second set of switches. The circuit further comprises a first power source and a second power source that are coupled to the first and second half-bridge power stages for supplying a first direct current (DC) voltage and a second DC voltage, respectively. The circuit further comprises an output interface coupled to the first and second half-bridge power stages and configured to couple to a speaker load. The first and second half-bridge power stages are configured to generate a five-level differential signal for driving the speaker load. Each set of switches includes a respective first subset of switches configured to operate at a first voltage level and a respective second subset of switches configured to operate at a second voltage level.

A method includes determining whether an audio signal processed in a class-D amplifier is clipping. Responsive to clipping of the audio signal, a speaker voltage is estimated as a ratio of a power supply voltage of the class-D amplifier to a resistance of a speaker driven by the class-D amplifier. An amplitude of the audio signal is controlled based on the speaker voltage.

Provided are an audio playing module and an audio processing circuit thereof, and an electronic device, which belong to the technical field of electronics. In the audio processing circuit, by adding a feedback control circuit between a power circuit and an audio amplifier, the feedback control circuit can control, according to a change in the amplitude of an audio signal received by the audio amplifier, the power circuit to adjust a power supply voltage provided for the audio amplifier. Thus, when the amplitude of the audio signal changes, an appropriate power supply voltage can be matched for the audio amplifier by means of the feedback control circuit, thereby solving the problem of the audio playing module generating whistling when the amplitude of the audio signal changes, and thus optimizing the sound quality of played sounds.

Signal processing circuitry configured to receive an input signal and to output a processed output signal, wherein the signal processing circuitry is configured to: receive an indication of a temporal location of a transient in the input signal; and provide, in the processed output signal, a masking signal bridging the temporal location of the transient to mask the transient.

The invention relates to a low-power-consumption high-performance D-type power amplifier modulation method and circuit and a D-type audio power amplifier, and the method comprises the steps: separating a pair of BD mode modulated signals outputted by a PWM modulation module in the D-type audio power amplifier circuit into a common mode component and a differential mode component, and outputting the common mode component and the differential mode component; feeding back the common-mode component and the differential-mode component to a differential input end through a common-mode feedback link and a differential-mode feedback link respectively; wherein the common-mode component and the common-mode feedback link are completely arranged in the chip; the differential mode component and the differential mode feedback link are realized by a power MOSFET (Metal-Oxide-Semiconductor Field Effect Transistor) integrated inside the chip or an external power MOSFET; and outputting the differential mode component to a to-be-driven object through the power output stage. According to the method, after two feedback links are separated, in a zero-input mute state, no common-mode signal is sent to an output stage for amplification, and power consumption is not caused at the output stage, so that the overall efficiency is greatly improved; and meanwhile, static bottom noise is almost eliminated.

The invention provides an intelligent audio power amplifier system and a working method thereof. The working method comprises the steps that a dynamic noise suppression module enables an input signal after reverse noise injection to enter a distortion compensation module; a distortion compensation signal is synthesized by using a distortion compensation module; the life cycle parameter self-calibration module calibrates parameters and inputs the parameters to the dynamic noise suppression module and the distortion compensation module; the dynamic noise suppression module is based on an intelligent learning method and adopts a frequency band channel network, a time sequence channel network and a fusion layer; the life cycle parameter self-calibration module provides the temperature and the working time for the dynamic noise suppression module to optimize the time domain channel network and provides the temperature and the working time for the distortion compensation module to update the predistortion coefficient of the dynamic nonlinear distortion model; the output of the dynamic noise suppression module is provided for the distortion compensation module; and the output of the distortion compensation module is fed back to the dynamic noise suppression module. The problems of high-frequency noise interference, nonlinear distortion degradation and insufficient long-term stability are solved at the same time.

In an embodiment, a device includes: phase frequency detector (PFD) circuitry having an input and an output; an oscillator having an input and an output, the input of the oscillator coupled to the output of the PFD circuitry; clock divider circuitry having a first input, a second input, and an output, the first input of the clock divider circuitry coupled to the output of the oscillator, the output of the clock divider circuitry coupled to the input of the PFD circuitry; and a controller having an output coupled to the second input of the clock divider circuitry.

A lavalier microphone has improved dynamic range. The lavalier microphone incorporates a pair of microphone transducer elements into the same housing or attached housings, with one of the microphone transducer elements being more sensitive to acoustic input than the other. The voltage signals from the respective transducer elements are processed and combined, resulting in a digital output audio signal with improved signal to noise both at lower and higher input acoustic levels.

The utility model discloses a discharge detection device for a dry-type transformer, and the device comprises a sound collection module which is used for collecting sound signals generated during the discharge of the dry-type transformer; the audio pre-amplification module is connected with the sound collection module and is used for carrying out primary amplification on the sound signal to obtain a first sound signal; the frequency selection module is connected with the audio pre-amplification module and is used for screening the frequency of the first sound signal to obtain a second sound signal; the audio amplification module is connected with the frequency selection module and is used for carrying out secondary amplification on the second sound signal to obtain a third sound signal; and the audio output module is connected with the audio amplification module and is used for outputting the third sound signal. According to the utility model, the sound collection module is arranged to collect sound during operation of the transformer, the audio pre-amplification module and the audio amplification module are adopted to amplify the sound to a frequency band which can be heard by people, the sound is output through the audio output module, and an operator hears the sound to judge the discharge condition of the transformer.

Apparatus and techniques for adaptively adjusting an input current limit of a boost converter that supplies power to a load, such as an amplifier. An exemplary circuit for supplying power generally includes a boost converter having an output coupled to the load, and logic configured to adaptively adjust the input current limit of the boost converter and apply the input current limit to the boost converter based on an estimated output power of the boost converter. One exemplary method for supplying power generally includes converting an input voltage to an output voltage by a boost converter to power a load for the boost converter, adaptively adjusting the input current limit of the boost converter based on the estimated output power of the boost converter, and applying the input current limit to the boost converter during conversion.

An example apparatus includes class D amplifier circuitry (130) having a first input, a second input, a third input, and an output, the first input of the class D amplifier circuitry (130) being coupled to the output of the class D amplifier circuitry (130); and class AB amplifier circuitry (140) having a first input, a second input, a third input, and an output, the first input of the class AB amplifier circuitry (140) being coupled to the first input of the class D amplifier circuitry (130) and the output of the class D amplifier circuitry (130), the second and third inputs of the class AB amplifier circuitry (140) are coupled to the second and third inputs of the class D amplifier circuitry (130) and the output of the class AB amplifier circuitry (140).

The invention discloses an overload protection method for an audio power amplifier, and relates to the field of audio power amplifiers. According to the invention, the instantaneous output amplitude signal of the power amplifier is divided by the preset overload threshold impedance to obtain the reference current dynamically changing along with the signal, and the actual output current is compared with the reference current in real time, so that the protection criterion is converted from a fixed current threshold to an adaptive impedance threshold; therefore, real-time identification and turn-off of a low-impedance abnormal load are realized, and normal output of a high-peak-factor audio signal is compatible.

The proposed solution relates to a computer-implemented method for outputting an amplified audio output signal (S2) using an amplification unit (1) for an acoustic vehicle warning system (100) to generate sound audible from persons in the vicinity of a vehicle (200). The method involves receiving an audio input signal (S1) and a temperature value (T) from a temperature sensing unit (20). A control unit (10) of the amplification unit (1) adjusts the amplification of the audio input signal (S1), taking into account at least one temperature reduction measure depending on the temperature value (T), in order to generate the audio output signal (S2). According to the proposed solution, the temperature reduction measure comprises at least one of i.a. adjusting the audio input signal (S1) with respect to at least one frequency (F1) of the audio input signal (S1) while maintaining or increasing the gain of at least one other frequency (F2) of the audio input signal (S1), ii. reducing an operating frequency of the amplification unit (1), and iii. adding at least one frequency (F3) to the audio output signal (S2) that is not included in the audio input signal (S1).

A class D amplifier including a pulse width modulation (PWM) control loop, an H-bridge driver, a PWM controller having a PWM pulse generator to generate PWM pulses supplied to the H-bridge driver in response to an ADC value output by an analog-to-digital converter of the PWM control loop, and a comparator. The comparator is configured to trigger a cutoff of a PWM pulse generated by the PWM pulse generator during a respective loop cycle of the PWM control loop and to trigger generation of at least one counter PWM pulse with an opposite polarity in the respective loop cycle such that a net pulse energy of the cutoff PWM pulse and of the at least one generated counter PWM pulse corresponds to a pulse energy representing the ADC value provided by the analog-to-digital converter of the PWM control loop during the respective loop cycle.

The invention discloses an audio amplification system based on phase reversal double-class-D power amplifier low power stress. The audio amplification system comprises a signal receiving and decoding module, an operational amplifier module, a double-group class-D power amplifier module, a filtering output module and a power supply module for supplying power to each module, which are connected with one another, a first operational amplifier unit in the operational amplifier module is configured to perform in-phase amplification on the analog audio signal; the second operational amplifier unit is configured to perform inverting amplification on the input signal; a first D-type power amplifier unit in a double-group D-type power amplifier module receives an in-phase signal and outputs an in-phase PWM (Pulse Width Modulation) power signal, and a second D-type power amplifier unit receives an inverted signal and outputs an inverted PWM power signal; and the first D-type power amplifier unit and the second D-type power amplifier unit work based on driving signals with opposite phases, so that peak current drawn from the power supply module is staggered or complementary in time to reduce instantaneous current stress and voltage fluctuation of the power supply module. The same audio output power is realized through ingenious signal phase management on the premise that the power supply cost is not increased or even reduced.

This application relates to methods and apparatus for controlling a switching driver to drive a load with a drive signal based on an input signal. A controller is configured to control modulation of at least one driver output node between selected switching voltages of a set of at least four different switching voltages to generate the drive signal. The output node is modulated according to a switching pattern and the controller is configured such that, for a given level of drive signal, the controller can selectively operate with a plurality of different switching patterns and selects an appropriate switching pattern to control at least one parameter of the switching driver other than the drive voltage.

A differential delta-sigma-modulator has an integrator (49) including a pair of single-ended amplifiers (46, 47). The differential delta-sigma-modulator comprises a sample clock (50) driving a first switchable capacitor configuration (31) and a second switchable capacitor configuration (32) at a predetermined switching cycle. The second switchable capacitor configuration (32) is adapted for sampling respective outputs from the pair of single-ended amplifiers (46, 47) on a pair of output sampling capacitors (C3, C5) in the first part of the switching cycle (P1), and charging a common mode capacitor (C4) with the average voltage of the voltage sampled by the pair of output sampling capacitors (C3, C5) in the second part of the switching cycle (P2). The voltage across the common mode capacitor (C4) represents the common mode voltage for the integrator (49).

A method for use with an acoustic system (referred to as a plant in control theory) (125) including a loudspeaker set (120), the method comprising obtaining an input audio sample (r n ) and a measured current (m n-d ) drawn by the loudspeaker set, and for a current time step, using a state space model of the loudspeaker set to generate a state vector (x n ) and a model-based estimated current (y n ), generating a predicted current (y n ) for a next time step based on the state vector (x e,n+1 ), generating a replacement audio sample (u n ) based on the state vector (x L,n ) for the current time step, generating a feedback signal (f n-d ) that contributes to the generation of the state vector (x n-d ) for a next time step based on the measured current (m n+1 ) and a time-aligned model-based estimated current (y n ), and generating an output audio sample (u n ) for the current time step, wherein when the predicted current (y e,n+1 ) is higher in amplitude than a threshold (I max ), the output audio sample (u n ) is either the input audio sample (r n ) or the replacement audio sample (u L,n ) for the current time step.

An audio signal processing method, and an audio signal processing device for executing the audio signal processing method includes, obtaining an audio signal and compressing the audio signal to generate a first compressed signal; calculating a duration for which a power value of the first compressed signal equals a first power threshold; determining whether the duration exceeds a first time threshold; when the duration exceeds the first time threshold, compressing the first compressed signal to generate a second compressed signal, and amplifying and outputting the second compressed signal.

A valve amplifier (105, 305) comprising an output transformer (110, 310) and a capacitor (210, 410) connected in parallel with a secondary winding (205, 405) of the output transformer (110, 310) Systems (100, 300), methods, and valve amplifier accessories (325, 500, 600) are also provided.

In some examples, a circuit include a class-D amplifier configured to receive an input signal and provide a drive signal based on and responsive to the input signal for driving an output device. The circuit also includes a current sensing circuit coupled to the class-D amplifier and configured to provide voltage signals representative of current flow through the class-D amplifier. The circuit also includes a current measurement circuit coupled to the current sensing circuit and configured to provide a first and second voltage difference signals derived from the voltage signals. The circuit also includes a current summing circuit coupled to the current measurement circuit and configured to determine a first current signal representative of the first voltage difference signal, determine a second current signal representative of the second voltage difference signal, and combine the first and second current signals to form a current sum signal. The circuit also includes a converter circuit coupled to the current summing circuit and configured to convert the current sum signal to a voltage representation.

This invention provides an amplifier with improved bias tracking performance based on a DSP chip. It includes a file acquisition module that directly acquires the audio file to be played via an interface. The audio file is pre-input into the DSP for standard waveform analysis, which speeds up processing and reduces latency during actual playback. During actual playback, the actual played energy is compared with the pre-calculated energy to determine if it matches the pre-designed output. If it matches, the pre-designed output is used directly; otherwise, it is further input into a model for processing, ensuring operational stability and safety.

A method for operating a speaker, in particular a dynamic-speaker, in which a useful signal for producing sound by the speaker is linearized. According to the disclosure, the useful signal is filtered depending on frequency and only a filtered portion of the useful signal is linearized. Advantageously, the useful signal is filtered by a high-pass filter and / or a low-pass filter. In one embodiment, only a portion of the useful signal that is output by the low-pass filter is linearized.

An apparatus comprising; an amplifier (308) having an amplifier input and an amplifier output; a power stage (PSI) having a power stage input and a power stage output, the power stage input coupled to the amplifier output; and an overcurrent protection circuit (402, 404) coupled to the power stage, the overcurrent protection circuit having an overcurrent threshold control input coupled to the amplifier.

System for a real-time radio receiver network, wherein the system comprises: a remote digital antenna digital receiver (“RDADR”) with a processor; a first remote digital antenna (“RDA”) comprising a processor, at least one receiver and at least one antenna; a second RDA with a processor, at least one receiver and at least one antenna, an audio source that is coupled to the transmitter; and an audio output device coupled with the RDADR, where the RDADR, the first RDA and the second RDA are coupled to each other via a digital bus, wherein the first RDA and the second RDA attempt to receive one or more digital signals from a transmitter, and if the one or more digital signals are received without error by the first RDA or the second RDA, the RDA that has received the one or more error-free digital signals transmits the one or more error-free digital signals via the digital bus to the RDADR, where the audio source generates one or more digital signals, and where the audio source is at least a microphone or a musical instrument.

Provided are feedforward amplifier circuit, audio amplifier and audio playing device, including: a main amplifier circuit, an adder circuit, a sub-amplifier circuit and a subtractor circuit, wherein the main amplifier circuit adds distortion to a first input signal to output a distorted signal, and inputs the distorted signal to the adder circuit and the subtractor circuit for performing feedforward correction; the sub-amplifier circuit amplifies a second input signal and serves as a positive input of the subtractor, so that a signal gain output by the subtractor is equal to a signal gain of the main amplifier; the subtractor circuit, with the distorted signal as a negative input, reversely amplifies the distorted signal and afterwards performs feedforward; and the adder circuit superimposes the distorted signal output by the main amplifier circuit and the reversely amplified distorted signal output by the subtractor, so as to output a distortion-free signal.

Methods, systems, and computer program products are presented herein for rendering synchronous audio using audio rendering devices. A networked audio rendering device may include a first wireless communications module, a second wireless communications module, a microcontroller, at least one speaker driver, and at least one digital audio amplifier. Audio data may be received by a first wireless communications module. Timecode data may be received by a second wireless communications module operating at a sub-GHz ISM band. The audio data may be received by a microcontroller. The audio data is processed by the microcontroller to generate output audio signals based on the timecode data. The output audio signals may be amplified by at least one digital audio amplifier. At least one speaker driver may be driven, by the at least one digital audio amplifier, to render an audio output.

The application belongs to the technical field of integrated circuits, and specifically discloses a digital control programmable gain amplifier chip, which adopts an SIP packaged operational amplifier, a digital potentiometer, an input end Vin and an output end Vout, an alternating current signal is input from the Vin end, is connected to the inverting input end of the operational amplifier through a capacitor C1 and an input resistor R1, a positive power supply pin of the operational amplifier is connected to a positive electrode VDD of a power supply, a negative power supply pin is connected to the ground, a same-phase input end obtains a VDD / 2 direct current bias voltage through a precision resistor voltage dividing network and is connected after low-pass filtering, a high end of the digital potentiometer is connected to the Vout end, a low end is connected to the inverting input end of the operational amplifier, and a feedback path is formed through a sliding end. The operational amplifier and the digital potentiometer are integrated, high integration and modularization of the programmable gain amplification circuit are realized, and the problems of insufficient gain precision, decreased reliability of a multi-element system and large circuit volume caused by the on-resistance of an analog switch in a traditional discrete scheme are solved.

A switching circuit outputs a modulated signal. A low-pass filter circuit receives the modulated signal and applies low-pass filtering processing at a low-pass cut-off frequency to generate a demodulated signal to a load impedance. A replicating circuit receives the modulated signal and applies a replica of the low-pass filtering processing of the low-pass filter circuit to generate a reference signal. A signal processing circuit applies signal processing to the demodulated signal and the reference signal to generate at least one indicator signal which is indicative of a variation of the low-pass cut-off frequency.