57results about How to "Avoid sound quality" patented technology

The present invention relates to a flat panel audio output apparatus including a flat panel with a plurality of channel areas divided according to audio channels, with a division area separating the channel areas from each other to prevent vibrations between the channel areas from interfering with each other, and a plurality of vibrators provided in the respective channel areas, vibrating the respective channel areas to output sound based on audio signals provided to the vibrators corresponding to the audio channels. Thus, embodiments of the present invention provide a flat panel audio output apparatus and method, a video / audio output apparatus, and a flat panel audio output method, preventing sound quality from being deteriorated due to interference between sound output from a plurality of audio channels.

An image display device has a main display body provided with a thin display unit, a low-frequency loudspeaker (woofer) and two middle and high frequency loudspeakers (tweeters) and a stand supporting the main display body. The stand has a base being just opposite at its top surface to a duct outlet of the woofer, thereby sound emitted from the duct can be constantly absorbed by and reflected from the base. A housing containing the first loudspeaker and a pair of the second loudspeakers has a plurality of through holes formed in a first area corresponding to an aperture of each loudspeaker and a plurality of blind holes formed in a second area surrounding the first area, wherein the second area for the first loudspeaker is different in peripheral shape and / or size from the second area for each of the second pair of loudspeakers.

An exciter for directly vibrating a board treated for dustproofing and heat dissipation, provided with a vibrator comprised of an outer yoke, a permanent magnet, and inner yoke, having a coil fixed inserted between the outer yoke and inner yoke, and having a coupler member for coupling the exciter to the surface of the board. The coupler member and vibrator are connected by an elastic member. A closed space is formed in the exciter by providing a sealing member between the outer yoke and coupler member. A through hole is formed in the outer yoke. Due to the vibration-of the vibrator at the time of coil excitation, air convection is caused through the through hole between the closed space and the outside space.

Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

The invention provides an electronic device with high water resistance and good sound quality recording and playback functions. The digital camera includes: a first cover (31) having a first sound hole (21) formed at a position facing the microphone (25); a waterproof sheet (40) arranged to cover the first sound hole (21) ; a second cover (32) disposed separately from the waterproof sheet (40); and a second sound hole (22) formed in the first cover (31). The second sound hole (22) is dislocated on the outer periphery of the first sound hole (21), so that it is not located on the extension line of the axis direction of the first sound hole (21). The outer peripheral portion (40a) of the waterproof sheet (40) extends to a position facing the second sound hole (22).

The invention discloses an audio frequency device with a graphene composite sound film. The audio frequency device is provided with a vibration sound film, a shell frame, a drive circuit or a sensing circuit, wherein the shell frame defines a cavity in which the vibration sound film vibrates, and the drive circuit or the sensing circuit is used for driving the sound film to produce sound through vibration or sensing the vibration of the vibration sound film so as to output a signal. The vibration sound film of the audio frequency device is a composite sound film and is formed by compositing a fiber meshy graphene layer and a thin polymer film. By means of the graphene composite sound film, the thickness and quality of the composite sound film can be remarkably reduced, the acoustical frequency response characteristic of the sound film is improved, and the sensitivity and fidelity of the audio frequency device are improved. The audio frequency device is convenient to install, high in durability and reliability and low in manufacture cost, and tone quality and tone are greatly improved on the premise that high frequency response characteristic is kept.

The invention discloses a video playing method and device. The method comprises the following steps: acquiring viewer information, and judging whether juveniles exist in viewers or not according to the viewer information; when the juveniles exist in viewers, judging whether content to be played of a video includes X-rated content or not; and if the X-rated content is included, processing and playing the X-rated content. Through adoption of the video playing method and device, the X-rated content in the video can be processed flexibly according to the ages of the viewers, so that the influence of the X-rated content on juvenile audiences is avoided on the premise of ensuring the integrity of the watched video.

An earphone device converts electric signals to audible sound. The device includes an outer area configured to receive a power source. A plug area adjacent to the outer area may be configured to fit within a user's auditory canal. A dynamic transducer reproduces a predetermined frequency spectrum for the user. A sound channel terminating at an output of the dynamic transducer encloses the second transducer.

A buffer controller has a buffer for holding plural sets of data contained in a packet entered from a telecommunications network, a codec converter and a controller. When receiving a packet, the buffer controller has the controller put data, in the packet, in a storage position in the buffer corresponding to the sequence number of the packet, and makes a decision as to whether or not the codec conversion is to be performed. If packets are out of sequence, lost or dropped during communication, the buffer controller can correct the packet order and compensate the packet loss with the minimum delay.

The present invention disclosed an all-digital speaker system device based on multi-bit Σ-Δ Modulation, comprising an A / D converter, an interpolation filter, a Σ-Δ modulator, a dynamic mismatch regulator, a differential buffer and a speaker array. The present invention takes advantage of the Σ-Δ modulation technique to effectively reduce the cost and complexity of the hardware realization of the speaker system device, thus realizing the full digitalization of the whole sound transmission link and the integration of the system device with low power dissipation and volume. Furthermore, the system device according to the present invention is provided with a better control ability on the local sound field, thus providing a better practicable method for the voice secret transmission.

A resistor 1 is used in high-fidelity amplifiers for audio equipment. The resistor 1 includes a cylindrical resistor body 2, a tubular sheath 3 into which the resistor body 2 is coaxially inserted, and a conductive film portion 31a formed on the inside surface 3a of the sheath 3. The conductive film portion 31a faces to a resistance film 22 covering the surface of the resistor body 2 with an annular space 6. An insulating slit 32 is formed at a central point along the resistor axis 1a, separating the conductive film 31 into left and right parts that are electrically isolated from each other. The sheath covering the resistor body 2 prevents distortion of signals in the resistance film 22 caused by extraneous electrostatic induction charges.

The invention discloses a motor cover for a dust collector. The motor cover is arranged inside a body of the dust collector, is fixed with a motor of the dust collector and guides the flowing direction of air exhausted by the motor. The motor cover comprises a shell and an exhaust outlet, wherein the shell is combined with a shell cover to form the motor cover, surrounds the motor and forms an enclosed exhaust space around the motor; the exhaust outlet is formed on the shell and is communicated with spaces inside and outside the motor cover to directionally exhaust the air inside the exhaust space; a sleeve is arranged inside the shell around the motor and comprises an externally sealed back plate and a noise-reducing plate arranged on the inner side of the back plate; a space is respectively reserved among the noise-reducing plate, the motor and the back plate, and the noise-reducing plate and the back plate form a whole; a plurality of through holes are formed on the noise-reducing plate; and exhaust holes are formed on the sleeve. By changing the thickness of the noise-reducing plate, diameters of the through holes formed on the noise-reducing plate, porosity and other parameters, a specific part in noise is pertinently weakened and eliminated; therefore, the sound quality of the noise is changed.

An apparatus and a method for transmitting audio signals in such a manner that audio signals transmitted and received in a mobile telephone network are preprocessed before being inputted to a voice encoder are provided. The audio signals are preprocessed by using an optimal filter gain based on error signals obtained when the audio signals are preprocessed and outputted by the voice encoder and the synthesizer. Therefore, the sound quality of the audio signals is hardly degraded by the voice encoder.

The invention discloses a COMSOL software-based loudspeaker frequency curve analysis method and device. The method comprises the following steps of: establishing a 2D or 3D loudspeaker simulation model and endowing the loudspeaker simulation model with material characteristics; carrying out finite element analysis on the loudspeaker simulation model; judging whether a frequency response curve of the loudspeaker simulation model accords with a design requirement or not; if the judging result is positive, taking structure parameters and material parameters as design parameters of a loudspeaker; and if the judging result is negative, adjusting the structure parameters and the material parameters until the frequency response curve accords with the design requirement. According to the method and device disclosed by the invention, acoustic-structure coupling analysis is carried out on the loudspeaker simulation model on the basis of COMSOL software, so that peak values and valley points on the loudspeaker frequency response curve can be decreased, thereby improving the tone quality and performance, avoiding the problem that the tone quality and parameters of the loudspeaker are unqualified as the loudspeaker is designed according to rules of thumb, improving the production efficiency and decreasing the production cost.

A method for interrupting a conversation transmitting terminal in a narrow-band wireless digital communication system is performed based on a 2-time slot TDMA (Time Division Multiple Access) system and comprises the following steps: (1) presetting a plurality of synchronization words of interrupt requests with priority levels in each voice superframe, wherein the each voice superframe consists ofsix voice frames, i.e., an A frame, a B frame, a C frame, a D frame, an E frame and an F frame, the middle of the A frame is a voice synchronization word, the synchronization words of the interrupt requests are arranged between the E frame and the F frame; (2) when a transmitting terminal transmits voice and a receiving terminal waits to receive the voice transmitted by the transmitting terminal,switching to a receiving mode at one time slot after the E frame of the each voice superframe and trying to decode and match the synchronization words of the interrupt requests, and switching back toa transmitting mode when one time slot after the E frame is near the end no matter whether the synchronization words of the interrupt requests are matched successfully or not; and (3) interrupting thecalling of the transmitting terminal to perform normal transmission if a voice or data service is initiated when the receiving terminal is in a receiving state.

The invention relates to a case for an audio or video player having a front cover and a back cover, each having an inside wall and an outside wall. A speaker is placed between the inside and outside walls of the front cover, a cord is placed between the inside walls for connecting the speaker to the player through the inside wall of the front cover, and the speaker includes an output facing toward the outside wall. The speaker facing outwards permits the player to be heard when the case is closed without sacrificing sound quality.

Through the present invention, a person does not feel discomfort when touching a speaker-incorporating headrest, the speaker is arranged near the person, whereby the headrest is formed having a small size, and the sound pressure or sound quality of sound from the speaker is prevented from decreasing. The present invention has a plurality of speakers 25a, 25b, fiber pads 15 provided in front of the speakers 25a, 25b, a sound-insulating pad 16 provided between the fiber pads 15, and a cover 9 for covering the speakers 25a, 25b / fiber pads 15 / sound-insulating pad 16, the fiber pads 15 are formed from an elastic material having an internal three-dimensional network structure, and the sound-insulating pad 16 is formed from an elastic material for suppressing transmission of sound.

The invention provides a method for realizing digital audio encoding by using a new psychoacoustic model and quick bit allocation. The audio encoding method comprises the following steps of: by using a new psychoacoustic model, determining window types according to the variance of input audio signals; allocating quantized noise according to the determined window types and the energy of the input audio signals, namely calculating threshold values; performing quick bit allocation, giving up the conventional method of performing rate-distortion control by two layers of nested loops, taking a value of a global scale factor of the last frame as an initial value for global scale factors according to the short-time stable properties of the audio signals, and acquiring accurate an initial valve for each section of scale factor through model deduction; and finally acquiring values of a final global scale factor and each section of scale factor through fine adjustment. The method has the characteristics of low computational complexity and unconspicuous reduction of perceived tone quality.

An edge for a speaker making it possible to decrease a thickness of the speaker and achieve cost reduction by using no damper without lowering a sound quality is provided. The edge of a speaker is in a sheet-like and ring form and includes an outer fixing part formed on its outer peripheral part for fixing the edge to the frame (supporting member), an inner fixing part formed on its inner peripheral part for fixing the edge to the diaphragm, and a sheet-like part lying between the inner fixing part and the outer fixing part (connecting the inner fixing part and the outer fixing part). The respective convex portions are formed at least on both surfaces of the sheet-like part at the boundary portion of the sheet-like part and the outer fixing part, thereby forming the control parts for controlling the vibration of the diaphragm.

The application provides a method and a device for adjusting Bluetooth A2DP (Advanced Distribution Profile) coding setting. The method for adjusting Bluetooth A2DP coding setting provided by the invention, which is applied to a source, includes the following steps: when a Wi-Fi module state change event occurs in an initial state of an AVDTP (Audio / video Distribution Transport Protocol) link or after initialization of the AVDTP link, determining a coding setting matching the current state parameters of the Wi-Fi module of the source for the source; determining an enabling strategy of the coding setting according to the current state of the AVDTP link; and using the enabling strategy to enable the coding setting, and when the coding setting is enabled, sending the coding setting to a sink. The method and the device for adjusting Bluetooth A2DP coding setting provided by the application can ensure that the coding setting of the source can always adapt to Wi-Fi interference, in order to reduce the interference of Wi-Fi signals with transmitted data and to avoid sound quality problems caused by improper coding setting in the presence of Wi-Fi interference.

The invention relates to the technical field of sound boxes, in particular to a method for manufacturing a sound box net cover, a sound box net cover and a sound box. The manufacturing method of the speaker net cover is as follows: at least setting mist glue in the area of ​​the sound outlet on the front of the bracket and bonding the mesh cloth to the area of ​​the sound outlet. The mist glue will not block the sound hole, thereby avoiding the impact on the sound quality. Further, the mist glue makes the mesh and the bracket stick together at least at the sound hole area, and the mesh cloth will not hit the sound hole area even if it is loose, and thus will not vibrate, thus solving the problem caused by the mesh. The problem of abnormal sound caused by the impact of the cloth and the bracket. The speaker net cover is made by the manufacturing method of the above speaker net cover, and the speaker includes the above speaker net cover. The above-mentioned speaker net cover and speaker will not cause abnormal sound due to loose mesh hitting the bracket after being used for a certain period of time, and the sound quality will not be reduced due to loose mesh.

A language processing portion ( 31 ) analyzes a text from a dialogue processing section ( 20 ) and transforms the text to information on pronunciation and accent. A prosody generation portion ( 32 ) generates an intonation pattern according to a control signal from the dialogue processing section ( 20 ). A waveform DB ( 34 ) stores prerecorded waveform data together with pitch mark data imparted thereto. A waveform cutting portion ( 33 ) cuts desired pitch waveforms from the waveform DB ( 34 ). A phase operation portion ( 35 ) removes phase fluctuation by standardizing phase spectra of the pitch waveforms cut by the waveform cutting portion ( 33 ), and afterwards imparts phase fluctuation by diffusing only high phase components randomly according to the control signal from the dialogue processing section ( 20 ). The thus-produced pitch waveforms are placed at desired intervals and superimposed.

An audio circuit drives a speaker. A DSP corrects an audio signal S1 according to the state of the speaker. An amplifier drives the speaker according to a corrected audio signal S3.

In an input transistor of current mirror circuit, it is intended to prevent the current generated by Early effects from appearing in the output current of the current mirror circuit. Having an Early voltage detector 22 of same connection and structure as a driver circuit 21, and connected in parallel to the driver circuit 21, the current flowing in input transistor Q9 of the Early voltage detector 22 is detected. Output current (current IQ11) of Early voltage detector 22 is added to input current (current IQ5) of driver circuit 21. As a result, in input current (current IQ7) of current mirror circuit 25, current α due to Early voltage effects can be canceled. At the same time, current α due to Early voltage effects generated in input transistor Q5 of current mirror circuit 25 is prevented from appearing in output current (current IQ8) of current mirror circuit 25.

Provided are a method and an apparatus for sending multimedia data. A first device, which provides an audio signal to a second device, comprises: a control unit for dividing an audio signal inputted to the first device into a plurality of audio frames, comparing at least one previous audio frame pre-stored in a memory of the first device with a current audio frame among a plurality of audio frames, and selecting one from among the pre-stored previous audio frames on the basis of a similarity between the pre-stored previous audio frame and the current audio frame; and a communication unit for sending an identification value of the selected previous audio frame to the second device.

A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone is disclosed. Digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.