38results about How to "Improve output sound quality" patented technology

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 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.

A display apparatus including a display panel configured to display an image is disclosed. The display apparatus includes a supporting member on a rear surface of the display panel. The supporting member includes a groove. The display apparatus also includes a sound generator between the display panel and the supporting member, and configured to provide a vibration. The display apparatus further includes a vibration transfer member between the display panel and the sound generator, and configured to transfer the vibration of the sound generator to the display panel to generate sound.

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 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.

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.

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.

A display apparatus is disclosed. The display apparatus includes a display panel configured to display an image and a supporting member at a rear surface of the display panel. The display apparatus also includes a sound generator between the display panel and the supporting member, and at least one vibration member between the display panel and the supporting member.

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 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 is applicable to the technical field of loudspeakers. The invention provides a multi-segment dynamic range control method and system and a loudspeaker. The output power of the audio signal can be suppressed; filtering out low-frequency signals with over-low frequency; the amplification amount of an electric signal of an audio signal at at least one frequency point is adjusted; performing high-low frequency division on the audio signal; according to the loudspeaker, the high-frequency signal and the low-frequency signal after frequency division are dynamically limited respectively, so that the output power of the audio signal can be effectively suppressed, the audio signal is not distorted, the low-frequency signal with a larger dynamic range is filtered out, the sound breaking phenomenon is avoided, the multi-section dynamic range control of the audio signal is realized, the output sound quality of the loudspeaker is effectively improved, and the service life of the loudspeaker is effectively prolonged.

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).