39results about How to "Improve output sound quality" patented technology

An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

The invention belongs to the technical field of android audio playing, and particularly relates to a method and an apparatus for improving audio quality of an android device. The method for improving the audio quality of the android device comprises the following steps of obtaining a play sampling rate SR0 of a hardware abstraction layer of the android device; creating an audio stream instance according to the play sampling rate SR0; judging whether a sampling rate SR1 of a currently played audio is consistent with the play sampling rate SR0 or not; if the sampling rates are non-consistent, performing high-quality re-sampling play processing on the played audio; and if the sampling rates are consistent, performing play processing on the played audio. Through the method, self-defined high-quality re-sampling play processing in an android system can be realized, so that the problem of poor audio quality caused by adoption of a default re-sampling method of the android system is avoided.

An earphone structure including one or more composite chambers is described. Each of the composite chambers includes sub-speakers. The sounds generated by these sub-speakers are uniformly distributed in the composite chamber and a composite sound field is generated by these sub-speakers. The sound field generated by each of these sub-speakers can be adjusted by re-locating the position of the sub-speaker, for example, by re-locating the sub-speaker, or by re-locating the sub-speaker in different angles, to generate the composite sound field. The sound field generated by the main speaker and the composite sound field generated by these sub-speakers can form a spatial sound with very good quality. By using of its physical characteristics of locations of these sub-speakers, the sound generated by the earphone has a uniformly diffusion sound pressure and a surround effect, which allows a user of the earphone to enjoy a sound field similar to that in a theater.

A voice aware audio system and a method for a user wearing a headset to be aware of an outer sound environment while listening to music or any other audio source. An adjustable sound awareness zone gives the user the flexibility to avoid hearing far distant voices. The outer sound can be analyzed in a frequency domain to select an oscillating frequency candidate and in a time domain to determine if the oscillating frequency candidate is the signal of interest. If the signal directed to the outer sound is determined to be a signal of interest the outer sound is mixed with audio from the audio source.

A piezoelectric speaker includes a cabinet, piezoelectric vibrators horizontally mounted inside the cabinet at different elevations each having two eyed lugs respectively coupled to respective posts in the cabinet, a shock absorber coupled to the piezoelectric material of the piezoelectric vibrators, two conducting terminals accommodated inside the cabinet and clamped on the piezoelectric vibrators, and two clamps fastened to the conducting terminals to secure a respective signal line.

A portable electronic device including a first body, a second body, and at least one speaker is provided. The second body is pivotally connected to one side edge of the first body, such that the second body is selectively rotated to cover the first body, or to form an angle with respect to the first body. The speaker is movably disposed on an upper surface of the first body, and can be moved relative to the first body. Thus, the speaker can be moved and fixed to a predetermined position to achieve optimized output sound quality.

The invention provides an audio signal processing circuit which can eliminate audio signal noise and outputs audio signal with good audio quality. The audio-signal-processing circuit includes: a noise-detecting unit to detect presence or absence of noise in an audio signal generated based on an output from a tuner configured to receive a broadcast signal; a low-pass filter, having predetermined phase characteristics, to pass the audio signal having a band of frequencies lower than a predetermined frequency; a high-pass filter, having the predetermined phase characteristics, to pass the audio signal having a band of frequencies higher than the predetermined frequency; first- and second-output units to multiply the signals outputted from the low-pass and high-pass filters by first and second coefficients and output the multiplied signals, respectively; an adding unit to add the signals respectively outputted from the first and the second output units; and a coefficient control unit to, when the noise-detecting unit has detected the presence of noise, decrease the second coefficient below the first coefficient.

An earphone structure including one or more composite chambers is described. Each of the composite chambers includes sub-speakers. The sounds generated by these sub-speakers are uniformly distributed in the composite chamber and a composite sound field is generated by these sub-speakers. The sound field generated by each of these sub-speakers can be adjusted by re-locating the position of the sub-speaker, for example, by re-locating the sub-speaker, or by re-locating the sub-speaker in different angles, to generate the composite sound field. The sound field generated by the main speaker and the composite sound field generated by these sub-speakers can form a spatial sound with very good quality. By using of its physical characteristics of locations of these sub-speakers, the sound generated by the earphone has an uniformly diffusion sound pressure and a surround effect, which allows a user of the earphone to enjoy a sound field similar to that in a theater.

An acoustical phase inverter comprises a hollow cavity, at least one speaker connected with the cavity and an inverter tube which is connected with the cavity and communicated with the inside of the cavity, wherein the cavity contains a main body part and at least one extending part corresponding to the speaker; a sound cavity is formed in the main body part, a sound channel communicated with thesound cavity is formed in the main body part; the speaker is arranged on the tail of the extending part to seal the sound channel. The acoustical phase inverter has better low frequency response performance, ensures that the low frequency sound has lower degree of distortion, and has better output sound quality. In addition, the acoustical phase inverter has simple structure and smaller volume, and is convenient to carry.

The invention discloses a loudspeaker and electronic equipment. The loudspeaker comprises a shell, a vibration system and a magnetic circuit system; a first shell and a second shell of the shell are matched to define a mounting space, the first shell comprises a first extension part extending vertically and a second extension part extending horizontally, and the second shell comprises a third extension part extending vertically; in the projection in the vertical direction, the first extension part is located on the inner side of the third extension part, the end, away from the first extension part, of the second extension part is connected with the inner circumferential wall of the third extension part, the vibration system is connected with the bottom of the second extension part so as to be suspended in the installation space, and the vibration system comprises a vibrating diaphragm and a voice coil; the voice coil is at least provided with a first wire section and a second wire section which are oppositely arranged, and the magnetic circuit system is arranged in the installation space and is opposite to the vibration system at an interval. The invention aims to simplify the structure of the loudspeaker, effectively improve the sound production quality of the loudspeaker and improve the product performance of the loudspeaker.

A piezoelectric speaker includes a cabinet, piezoelectric vibrators horizontally mounted inside the cabinet at different elevations each having two eyed lugs respectively coupled to respective posts in the cabinet, a shock absorber coupled to the piezoelectric material of the piezoelectric vibrators, two conducting terminals accommodated inside the cabinet and clamped on the piezoelectric vibrators, and two clamps fastened to the conducting terminals to secure a respective signal line.

A wearable portable electronic device including a body and a speaker is provided. The body includes a base and a sidewall surrounding the base, wherein the sidewall has at least one opening. The speaker is disposed in the body, wherein a sound wave produced by the speaker is transmitted outward through the opening. An orthogonal projection of the speaker on the base is separated from an orthogonal projection of the opening on the base.

The embodiment of the invention discloses a noise-reduction circuit of an earphone microphone signal, a filter circuit and equipment. The noise-reduction circuit of the earphone microphone comprises a low-frequency filter module, a low and medium frequency filter module and a high frequency filter module, wherein the low frequency filter module is used for performing filter processing on the noise in a low frequency signal of the earphone microphone signal; a low and medium frequency filter module is used for performing filtering processing on the noise in the low and medium signal of the earphone microphone signal; the high frequency filter module is used for performing the filter processing on the noise in the high-frequency signal of the earphone microphone signal. By using the noise-reduction circuit of the earphone microphone signal, the filter circuit and the equipment disclosed by the embodiment of the invention, the filter noise reduction is performed on the noise in each frequency range through each of the low frequency filter module, the low and medium frequency filter module and the high frequency filter module, thereby achieving the level that the human ear cannot distinguish, and the output tone quality of the earphone microphone signal is improved.

A method for manufacturing a loudspeaker includes arranging, weaving, impregnating, drying, forming, cutting and assembling steps. In the arranging step, warp yarns and wires are arranged at intervals, and the wires are grouped as a wire group. In the weaving step, weft yarns are arranged at intervals and interwoven with the warp yarns and the wires to form a base material by weaving, and an area where the weft yarns are interwoven with the wires is defined as a wire disposing area. In the forming step, a wire damper is formed on the base material by thermoforming; two elastic adjustment areas are formed at the weft yarns between both sides of the wire group and the warp yarns closest to both sides of the wire group; and widths of the elastic adjustment areas are equal to each other and larger than distances between the remaining warp yarns.

The embodiment of the invention discloses a nonlinear compensation method for a loudspeaker. The nonlinear compensation method comprises the following steps: acquiring terminal voltage of the loudspeaker at the previous moment of the current moment; acquiring a state variable of the loudspeaker at the current moment according to the terminal voltage; acquiring a to-be-processed signal, modifying original linear parameters of the loudspeaker, and performing linear processing on the to-be-processed signal according to the modified linear parameters to acquire a linear processing signal; and performing nonlinear processing on the linear processing signal according to a nonlinear parameter of a loudspeaker and the state variable at the current moment to obtain a nonlinear signal, and outputting the nonlinear signal to the loudspeaker. The invention further discloses a nonlinear compensation system and device of the loudspeaker and a storage medium, the output tone quality of the loudspeaker can be effectively improved, and the service life of the loudspeaker is prolonged.

A portable electronic device including a first body, a second body, and at least one speaker is provided. The second body is pivotally connected to one side edge of the first body, such that the second body is selectively rotated to cover the first body, or to form an angle with respect to the first body. The speaker is movably disposed on an upper surface of the first body, and can be moved relative to the first body. Thus, the speaker can be moved and fixed to a predetermined position to achieve optimized output sound quality.

The invention brings forward a loudspeaker unit. The loudspeaker unit comprises a support, a vibrating body, a vibration apparatus and a control portion, wherein the end of the vibrating body is fixed at the inner side of the support, at least one opening is arranged at a position, which is deviated from the center of the vibrating body, in the vibrating body, the vibration apparatus is assembled on the vibrating body, the control portion is used for processing externally input signals and generating driving signals so as to drive the vibration apparatus, and the control portion comprises a processing portion, a DA converter and an amplifier which are successively connected. The loudspeaker unit related to by the invention can generate many minimal vibration modes and can enable frequency features to become relaxed and natural, thereby enabling output timbre to be good.

The invention discloses a tetrahedron truss suspension type electroacoustic device which comprises an electroacoustic unit and a regular tetrahedron truss. The electroacoustic unit is located in the regular tetrahedron truss and is connected to the four vertexes of the regular tetrahedron truss in a suspended mode through four elastic pieces. According to the electroacoustic device of the presentinvention, the structure is stable; the gravity center is low; due to vibration reduction and isolation effects of the elastic pieces, the device can be an ideal placing structure for the electroacoustic unit, by adopting the placing structure, the vibration of the electroacoustic unit does not influence other parts such as the regular tetrahedron truss, and the vibration of the other parts doesnot influence the electroacoustic unit, so that the sound output is not influenced by the outside, and the higher output sound quality can be ensured.

The invention discloses a vehicle-mounted anti-seismic audio horn, which comprises an audio shell, a buffer shell, an audio horn, a shock-absorbing column, an anti-seismic support, a fixed mounting plate, a shock-absorbing strip, and a shock-absorbing rubber surface. A buffer shell, the sound horn is fixedly installed in the buffer shell, and the top of the sound horn is provided with a plurality of shock-absorbing columns, and the tops of the shock-absorbing columns are fixedly connected with the top surface of the inner wall of the sound shell, and the top of the sound shell The rear side wall is provided with a plurality of anti-seismic supports, the bottom of the anti-seismic support is provided with a fixed mounting plate, the upper and lower bottom surfaces of the audio housing are provided with a plurality of shock absorbing strips, and the left and right outer surfaces of the audio housing are provided with Shock-absorbing rubber surface, compared with the prior art, the invention greatly increases the shock resistance of the car audio speaker, effectively reduces the impact of vehicle vibration on the speaker's stability, prolongs the service life of the car audio and improves the playback effect of the sound .

The invention relates to the technical field of headphones, in particular to a ceramic reverberation earphone. The ceramic reverberation headset includes an earphone housing and a sounding unit, characterized in that: the earphone housing is divided into a rear housing and a front cover, a sound amplifying sleeve is arranged in the rear case, and the sound generating unit is installed in the rearcase. The sound generating unit comprises a resonance unit, a resonance ring, a ceramic horn and a sound guiding film. The resonance ring is nested on the front side of the resonance unit. The front cover is provided with two sound outlet holes, the front cover is arc-shaped, and the mounting flange is arranged behind the front cover. The gain effect of the ceramic reverberation headset provided by the invention is as follows: as the ceramic horn is adopted, the absorption rate of the output sound quality can be greatly reduced, and the frequency of the output sound quality is improved by thecooperation of the resonant ring and the resonant unit, the echo sound effect is generated, and the sound is more directly transmitted to the eardrum by the front sound guide membrane and the sound guide cover, and the output sound quality of the earphone is improved in many aspects.

The invention discloses a pulse width modulation module, an audio power amplifier circuit and electronic equipment, the pulse width modulation module is applied to the audio power amplifier circuit, and the pulse width modulation module is used for receiving a control signal through a control end and selecting a corresponding data source according to the control signal to carry out pulse width modulation so as to output a pulse width modulation signal; and the data source comprises a first data source and a second data source, the first data source is a first data signal generated according to the control signal, and the second data source is a second data signal generated after a digital audio signal input by the audio power amplifier passes through a digital-to-analog converter. According to the pulse width modulation module, the data source selected by the pulse width modulation module is controlled to perform pulse width modulation according to the control signal, the first data source is generated according to the control signal, and the duty ratio of the pulse width modulation signal corresponding to the first data source is 0 or low level, so that the driven power amplifier stops working, the power consumption of the power amplifier is reduced, and noise is reduced.

The invention relates to a flat panel loudspeaker, a display device and display equipment. The flat panel loudspeaker comprises a plate structure, voice coils, voice coil frameworks and a magnetic field generating device. The number of the voice coils is multiple, and the multiple voice coils are concentrically arranged. The voice coils and the voice coil frameworks are equal in number and are in one-to-one correspondence; each voice coil is wound on the corresponding voice coil framework and is electrically connected with the magnetic field generating device; the voice coil framework corresponding to each voice coil is fixedly arranged on the plate structure; the plurality of voice coils are divided into a first voice coil and a second voice coil, and the number of the first voice coil is one; the first voice coil is used for exciting a first signal into sound, and the first signal is a signal of all frequency bands in a signal to be played; and each second voice coil is used for exciting a second signal into sound, and each second signal is a specific frequency band signal in the signals to be played.

The invention discloses a D-class audio power amplifier and an audio signal processing method thereof. The D-class audio power amplifier comprises a pre-amplification module, a triangular wave generator, an integrator, a feedback loop, a comparer, a driving module and an amplitude detector, wherein the amplitude detector is used for detecting an amplitude value of an input audio signal; and the amplitude value of an input voltage is in a range of 0 to 2 V. Because the amplitude detector is added, the D-class audio power amplifier reduces the saturation distortion of a restored audio signal. A pulse-width modulation (PWM) signal drives a loudspeaker by a power switch, and the loudspeaker restores the audio signal. Because the pulse width of the PWM signal accurately contains the amplitude information of the input audio signal, the D-class audio power amplifier also improves output tone quality.

The invention provides a compressor limiter. The compressor limiter comprises a gain adjustment module, a loudspeaker module, a clipping control module and a clipping amplitude generator. The gain adjustment module is used for outputting a first output signal, and feeding back and adjusting, according to the first output signal, a first signal gain for realizing compression and amplitude limiting of an input signal. The loudspeaker module is used for performing power amplification on the first output signal and outputting a second output signal. The clipping control module is used for feeding back and adjusting, according to clipping distortion occurring in the second output signal, a second signal gain for realizing power amplification of the first output signal. The clipping amplitude generator is used for automatically adjusting the limiting amplitude according to the clipping amplitude. Under the joint action of the first signal gain and the second signal gain, the compressor limiter controls the second output signal output by the compressor limiter within the limiting amplitude for further output, so that the clipping distortion occurring in the second output signal is eliminated automatically, the output tone quality of the loudspeaker module is improved, the user experience is improved, and the clipping amplitude and limiting amplitude are controlled in a relatively fixed proportion.

The invention provides a compression limiter, comprising a gain adjusting module, which outputs first output signals according to input signals of the compression limiter, and in accordance with the first output signal, carries out feedback regulation of first signal gains used for realizing signal compression limiting; a sound amplification module, which is connected with the gain adjusting module to carry out power amplification of the first output signals outputted by the gain adjusting module and to output second output signals; and a clipping control module, which is connected with the sound amplification module to carry out feedback regulation of second signal gains used for realizing power amplification of the first output signals according to clipping distortion of the second output signals outputted by the sound amplification module. Under the joint effect of the first signal gains and the second signal gains, the second output signals outputted by the compression limiter are outputted within the amplitude limit. The clipping distortion of the second output signals is automatically eliminated, thereby increasing the output sound quality of the sound amplification module and improving the user experience.

The invention relates to multi-channel audio signal processing, in particular to a method of processing a multi-channel audio signal and to a signal processing device. A method of processing a multi-channel audio signal is disclosed, comprising the steps of: receiving an input sum signal (s) representing a sum of a first audio signal and a second audio signal; receiving an input difference signal (d) representing a difference between the first and second audio signals; decorrelating the sum signal to provide a decorrelated sum signal (s d ); calculating a first gain (g s ) from a cross-correlation of the sum and difference signals (s,d) and the power of the sum signal; calculating a second gain (g sd ) from a cross-correlation of the sum and difference signals (s,d) and the power of the sum and difference signals; calculating an output difference signal (d') from a sum of the first gain (g s ) applied to the sum signal (s) and the second gain (g sd ) applied to the decorrelated sum signal (s d ); and providing an output stereo audio signal (l,r) from a combination of the output difference signal (d') and the input sum signal (s).

The invention relates to the technical field of voice signal processing, and particularly relates to a microphone-based regulation circuit and a microphone. An electrical signal acquired through an acquisition circuit is very weak. The electrical signal is amplified through a first amplifier. As the electrical signal through primary amplification still cannot drive a follow-up circuit, there is a need for secondary amplification of the electrical signal. In order to reduce attenuation of the high-frequency or low-frequency part of the electrical signal, before secondary amplification of the electrical signal, the unnecessary high-frequency or low-frequency component in the electrical signal is filtered through a regulation branch, and the electrical signal after filtering is amplified. Thus, the frequency domain response curve of the electrical signal is straight, and the quality of output sound of the microphone is improved. In the application, the frequency response of the high-frequency part of the microphone circuit is regulated through the regulation branch, so that the whole frequency response curve is 'dragged' straight, and the quality of output sound of the microphone is improved.