113results about How to "Improve listening effect" patented technology

The present invention relates to a method and system for audio encoding and decoding and a method for estimating a noise level, and the method for estimating a noise level in the present invention comprises: estimating a power spectrum of an audio signal to be encoded according to a frequency domain coefficient of the audio signal to be encoded; and estimating a noise level of a zero bit encoding subband audio signal according to the power spectrum obtained by calculating, and this noise level for controlling an energy proportion of noise filling to spectral band replication during decoding; wherein a zero bit encoding subband refers to an encoding subband of which allocated bit number is zero. The present invention can well reconstruct the uncoded frequency domain coefficients.

The invention relates to a self-adapting volume control method, a device and a communication terminal belonging to the communication technology field. The invention discloses the self-adapting volume control method comprising: receiving an audio signal from a sound source; acquiring sound source corresponding to the audio signal and an input, output gain parameters corresponding to the sound information; controlling an input, output volumes according to the input, output gain parameters. The invention also discloses the self-adapting volume control device comprising a receiving unit, an acquisition unit and a control unit. The invention also discloses the communication terminal comprising said device. The advantage of the invention is to automatically control the input, output volume and enable the user to obtain satisfactory listening effect.

The present invention relates to a method and device for spectral band replication, and a method and system for audio decoding, and the method for spectral band replication comprises: A. searching for the position of a certain tone of an audio signal in MDCT frequency domain coefficients; B. according to the tone position, determining a spectral band replication period which is a bandwidth from a 0 frequency point to a frequency point of tone position, and a source frequency segment which is a frequency segment from a frequency point of the 0 frequency point shifting copyband_offset frequency points backwards to a frequency point of the frequency point of the tone position shifting the copyband_offset frequency points backwards, wherein said offset copyband_offset is greater than or equal to 0; and C. according to the spectral band replication period, carrying out spectral band replication on zero bit encoding subbands.

The invention provides a system for processing an electronic cochlear signal and coding electric stimulation and a processing method thereof. The system comprises an acoustic signal analyzing and processing module, an electric stimulation resolution testing and processing module and an electric stimulation coding module, wherein the acoustic signal analyzing and processing module analyzes an acoustic signal and extracts characteristic information to be transmitted by low frequency and high frequency channels; the electric stimulation resolution testing and processing module measures and fits individual electric stimulation part resolving power and electric stimulation speed resolving power of an electronic cochlear user, acquires individual resolution of the electronic cochlear user and perceives parameters of an electric stimulation mode through calculation processing; and the electric stimulation coding module codetermines a discharge mode of an electronic cochlear implant electrode according to the ability of the electronic cochlear user for receiving the electric stimulation information and an acoustic signal processing result. The system can farthest promotes the hearing ability of the electronic cochlear user and improve the ability of the electronic cochlear user in speech recognition and understanding of tone language and music under a noise background.

The invention provides a method for monitoring digital audio signal volume. The method is achieved by combined action of hardware and programs. A digital audio signal volume monitoring card which comprises a digital audio acquisition unit, a decoding unit, a volume computing unit and a communication unit is adopted for the hardware. A computer accesses the volume monitoring card through the programs, and under the condition of not affecting other producing and playing normal operation, the volume of input audio signals and the dynamic range of the volume are recorded, displayed, stored and analyzed; the digital audio volume monitoring card acquires the digital audio signals and computes the corresponding volume of the audio signals; the digital audio acquisition unit is provided with three sets of input interfaces, and each set of the interface is provided with three input interface circuits suitable for digital audios with AES / EBU signals or S / PDIF signals, the input interface circuits are used for receiving the AES / EBU signals or the S / PDIF signals input in a balanced mode and imbalanced mode respectively; the volume computing unit computes the volume data result.

The invention relates to a medical instrument accessory, in particular to a stethophone accessory. The invention discloses a combined membrane which comprises a round membrane and a fixer. The fixer comprises three parts including a big ring, a small ring and a connecting part, wherein the big ring is arranged below the connecting part, the small ring is arranged above the connecting part and radially provided with a ring slot, and the outer edge of the round membrane is positioned in the ring slot. The invention also provides a stethoscope head which comprises a stethoscope head body and a cylinder, wherein the cylinder and the stethoscope head body are integrated into a whole, the end surface of the cylinder is provided with an axial ring slot and a sound collecting surface; and a soundguide hole and a fixing hole are arranged on the sound collecting surface. The membrane has the advantages of good restoration and accurate positioning; the stethoscope head is easy to install and hasstrong tightness; and users can acutely acquire a matching structure of multitone only by changing force applied on the stethoscope head body.

A wireless headset comprises a wireless transceiver and a speaker, the wireless transceiver is used for carrying out the communication with an audio device by the wireless transmission mode, and the speaker is used for outputting the voice. The wireless headset further comprises a register and a player, the register is used for storing an audio file received by the wireless transceiver, the player is used for playing the audio file in the register, and the speaker is used for outputting the voice which is corresponding to the audio file. The wireless headset receives and automatically plays the whole audio file which is sent by the audio device by the wireless transmission mode, thus avoiding the possible interference and distortion on audio signals by the wireless transmission and ensuring a user to obtain the better listening effect. The invention further provides the audio device and an audio playing method which are corresponding to the wireless headset.

An embodiment of the invention discloses a broadcast control method and terminal. The method includes the steps: detecting a first sliding track inputted on a device list interface; acquiring device identifiers in a device identifier area with the first sliding track passing by; generating a target device group comprising the device identifiers; receiving an inputted voice message according to the target device group, and analyzing the voice message to obtain voice duration and voice rate corresponding to the voice message; transmitting a broadcast instruction comprising the voice duration and the voice rate to devices corresponding to the device identifiers to control the devices to begin timing when the broadcast instruction is received, and broadcasting a song according to broadcast rate when timing duration reaches the voice duration. A plurality of broadcast devices can be grouped, the broadcast devices in the device group synchronously broadcast songs, and listening effects are improved.

The invention provides a prompt method and a mobile terminal. The mobile terminal comprises a display screen and an exciter, wherein the exciter is used for driving the display screen to make a sound.The method comprises the following steps that: determining a target area, wherein the target area is an area where the sound making parameter of the display screen conforms to a preset condition under the driving of the exciter; and in the target area, displaying prompt information, wherein the prompt information is used for prompting a user to display the area with the prompt information as thearea of which the sound making parameter conforms to a preset condition. Therefore, the prompt information used for prompting the user is displayed, and therefore, a hearing effect of the user for sound made by the display screen can be improved.

This invention relates to a kind of headset with aeration ear pad, including shell, speakers, speaker board, headband, linking handle, the speaker is fixed on the speaker board, the speaker board is fixed in the shell, its characteristic is that setting ring ear pad in the speaker board, installing lock seized location in the board, locking the ring ear pad in the seized location by pressure circle. The ear pad in this invention is made of nontoxic, good stability soft silicone and thermoplastic material, the structure is new, it also has the characteristic such as aeration, cool, sanitation, durable. Wearing long will not make the user feel pain pressure and stivy, it is a kind of headset that wearing comfortable, protecting the ears, listening with good effect, convenient, durable.

The invention discloses a gain adjustment method, which comprises the following steps of: obtaining an average power value of an audio file based on a time domain signal and a preset power value undereach level of volume; obtaining the average power value of the audio file based on the time domain signal; and obtaining the preset power value under each level of volume, calculating an expected gain value according to the average power value and the preset power value; calculating a preset power value according to the average power value, the expected gain value and a preset reference power value; Gain adjustment of the audio file is performed according to the preset power value. The invention also discloses an audio device and a computer-readable storage medium. The invention can avoid thedamage of the auditory system caused by the sudden increase of the loudness or the manual adjustment of the volume of the audio file by the user, thereby improving the auditory effect of the user.

The invention discloses a speech quality improving method and system and a mobile terminal. The method comprises: obtaining, by the mobile terminal, a spatial position from the face of a user to the mobile terminal; determining audio parameters corresponding to the spatial position from the face of the user to the mobile terminal; adjusting audio parameters of a speech processing module of the mobile terminal to the audio parameters corresponding to the spatial position from the face to the mobile terminal; and outputting a speech signal, which is received from a network side, after being processed by the speech processing module, the audio parameters of which are adjusted. By means of the method, the hands-free frequency response and sensitivity can basically be kept in a relatively good state and can be basically kept unchanged, thereby avoiding the case of low frequency or medium-high frequency loss of a frequency response curve due to change of the distance or the angle from the mobile terminal to the face, improving speech quality in video call, and improving the listening effect of the user.

The applicaiton provides a method and a system for outputting surround sound so as to solve the problem that the output surround sound is not accurate and thus the listening effect of a user is influenced. The method comprises the following steps: obtaining position coordinates of a sound image point; inputting the position coordinates of the sound image point to a neural network and determining a mapping matrix corresponding to the sound image point output by the neural network; receiving an input sound signal and determining the sound signal to be output according to the mapping matrix; and outputting the sound signal to be output to a corresponding microphone. According to the method and the system, the neural network is utilized for completing output of the surround sound so that the output surround sound is more accurate and the listening effect of the user is improved.

The invention provides a sound system which comprises a main sound device, a wireless sound device and a wireless headset device, wherein the main sound device comprises a sound input selection unit, a digital signal processing unit, a horn driving unit, a wireless transceiving unit, a control unit and a horn; the wireless round device comprises a first wireless transceiving unit, a first control unit, a first horn driving unit and a first horn; the wireless headset device comprises a second wireless transceiving unit, a second horn driving unit, a second control unit and a second horn; the second transceiving unit is used for receiving a wireless signal sent by the main sound device and enabling the second horn driving unit to drive the second horn; the second control unit is in charge of calculation and treatment; when the wireless headset device is started, a notification is sent to the main sound device, and the horn of the main sound device and the first horn of the wireless sound device are turned off, so as to enter a silent mode. The sound system can select a wireless headset mode or a sound mode.

The present invention relates to a three-channel holographic sound field playback method and a sound field collecting device. A multichannel sound source outputs a left sound channel signal, a right sound channel signal, and a front sound channel signal, wherein the output left sound channel signal and the right sound channel signal are directly input into a left sound channel speaker and a right sound channel speaker, respectively, which is characterized in that: the front sound channel signal output by the multichannel sound source is input into a front sound channel speaker after gain control.

The invention provides a laser interception device based on a correlation detection technology. The laser interception device based on the correlation detection technology comprises a focusing mirror, a modulation laser signal emission source, a modulation signal generator, a filter lens, a laser signal receiver, a correlation detector, a power amplifier and a loudspeaker, wherein an output end of the modulation signal generator is connected with the modulation laser signal emission source and provides laser modulation signals, the modulation laser signal emission source generates and emits modulation laser signals according to the laser modulation signals, the other output end of the modulation signal generator is connected with an input end of the correlation detector; the laser signal receiver receives reflected laser signals, the reflected laser signals are pre-processed and then input to the other input end of the correlation detector, an output end of the correlation detector is connected with an input end of the power amplifier, and an output end of the power amplifier is connected with the loudspeaker. By means of the laser interception device based on the correlation detection technology, the influence of external environment factors on the interception device can be remarkably reduced, the dynamic range of the receiver can be increased, the signal to noise ratio of interception signals can be increased, and audio interception effects are improved.

In an information processing method provided by the invention, after audio data information to be played is received, when an audio playing module is at a first state, the audio playing module comprises a first playing module and a second playing module, so the two playing modules play first audio information converted from the audio data information, and the audio information is played in a stereo mode; and when the audio playing module is at a second state, the audio playing module comprises the first playing module, so only one playing module is used for playing second audio information converted from the audio data information, and the audio information is played in a mono mode. In the method, with the combination of different states of the audio playing module, the audio data information to be played is converted into different types of audio information for playing, a user can select the corresponding playing mode according to a situation so as to control an electronic device to produce a mono or stereo sound, and listening effects are improved.

The invention relates to a DSP sound system and a sound system acoustic automatic test method, the DSP sound system comprises a microphone module, an execution module, an analog-to-digital conversion module, a DSP digital signal module, an MCU control module, a digital-to-analog conversion module, a power amplifier module and a loudspeaker module, the DSP digital signal processing module is used for testing a digital audio signal and forming a test result; the MCU control module outputs a control signal according to a test result and controls a sound system to adjust parameters, the digital-to-analog conversion module is used for converting a digital audio signal into an analog audio signal, the power amplifier module is used for amplifying the analog audio signal, and when the loudspeaker module receives the analog audio signal, the loudspeaker module plays sound. According to the invention, the acoustic test efficiency of the sound system is improved, the sound system has a better sound listening effect, and the acoustic test of the sound system can be conveniently completed by general users.

The invention relates to a medical instrument accessory, in particular to a stethoscope accessory, and particularly relates to a combined-type stethoscope membrane assembly. The combined-type stethoscope membrane assembly comprises a colloid-wrapped part and membrane, wherein the colloid-wrapped part comprises a first ring belt platform and a second ring belt platform. The diameter of the first ring belt platform is smaller than that of the second ring belt platform, a first annular lateral wall is arranged between the first ring belt platform and the second ring belt platform, a second annular lateral wall is arranged at the rim of the second ring belt platform and extends towards the direction away from the first ring belt platform, and the membrane is attached to the end surface outside the first ring belt platform. The stethoscope accessory is strong in anti-audio-frequency-damping ability and durability, reasonable in effective contact area, and capable of accurately acquiring moving information of internal organs of the body and ensuring that a user makes correct diagnosis decisions.

The invention provides a transcription repairing method of a video tape audio signal. The method comprises the following steps: using a magnetic tape surface coating repairing technology to process a video tape; transcribing the processed magnetic tape audio signal twice, to obtain an audio main file and an audio auxiliary file, the contents of the audio main file and the audio auxiliary file being the same; determining reference points for matching the audio main file and the audio auxiliary file, and according to the reference points, adjusting the audio main file or the audio auxiliary file; performing framing processing on the adjusted audio main file, to obtain a plurality of audio frames, and for each audio frame, searching a matched segment in the adjusted audio auxiliary file; based on a search result, determining audio frames missed in the audio main file, and using the corresponding matched segment in the audio auxiliary file to replace the missed audio frame, and splicing the audio frames, to repair the audio main file. The technical scheme solves problems of audio signal missing and distortion in an audio signal transcription process of an existing U-Matic video tape.

Disclosed is a digital audio loudness measuring card. The digital audio loudness measuring card is a measuring board card installed on the PCI express of a computer and comprises a digital audio collecting unit, a decoding unit, a loudness calculating unit and a communication unit, wherein the board card collects a digital audio signal directly and calculates the corresponding loudness of the audio signal, the loudness calculating unit comprises three sub-units, namely a filtering calculation unit, an integral average calculation unit and a weighting calculation unit, connected in sequence to obtain a loudness data result, the loudness data result of the loudness calculating unit is connected to the computer by the communication unit and then is recorded, displayed, stored and analyzed by the computer, the digital audio collecting unit comprises as many as three sets of input interfaces, each set of interfaces comprises three kinds of digital audio input interface circuits adapted to AES / EBU or S / PDIF signals, the three kinds of input interface circuits are used for receiving the AES / EBU signals or the S / PDIF signals input in a balanced mode or in an unbalanced mode, and selection of each set of digital audio input interface circuits is achieved through a multi-path selector switch.

The invention discloses an anti-coupling acoustic structure on a refrigerator. The anti-coupling acoustic structure on the refrigerator comprises an installing panel used for being connected with a refrigerator door body or a refrigerator body, and an acoustic body; a plurality of through holes are formed in the installing panel; the acoustic body is arranged on the back face side of the installing panel; a gap exists between the acoustic body and the back face of the installing panel; and a surrounding and blocking part is arranged in the gap. According to the anti-coupling acoustic structure on the refrigerator, the acoustic body and the installing panel are arranged in a spaced manner, electrostatic coupling is not prone to happening when an electromagnetic compatibility experiment is performed on the refrigerator, the experiment precision can be guaranteed easily, meanwhile, it is avoided that the acoustic body and the installing panel resonate in the using process, and consequently the tone quality is affected, the surrounding and blocking part has the sound gathering effect, the directivity of sounds made by the acoustic body can also be improved, and a better listening effect is acquired.

The invention discloses a double-earphone sound effect balance adjusting method and a system. The method comprises the following steps: taking the position of a noise source; defining one earphone closer to the noise source in the double earphones as a near-noise earphone, and defining the other earphone as a far-noise earphone; respectively obtaining the external sound collected by the near-noiseearphone and the far-noise earphone, and calculating the external volume decibel difference between the external sound of the near-noise earphone and the external sound of the far-noise earphone; andwhen the volume decibel difference of the sound outside the ear exceeds a preset first decibel threshold, increasing the output volume of the far-noise earphone, and / or reducing the output volume ofthe near-noise earphone. The invention provides the double-earphone sound effect balance adjusting method and the system, and the method and the system can automatically adjust the volume output values of a left earphone and a right earphone when a user generates noise in a side direction, so as to balance the volume values received by two ears, the earphone e listening effect is improved, and damage on the hearing of the user can be prevented.

A multifunctional bed with a foldable board for the integration of video, audio and camera functions is composed of main frame, back rest, mattress, legs, foldable board with LCD, operating panel, camera, loudspeaker, clock, lamp, telephone, etc.

The invention discloses a multi-channel wireless radio system which comprises a wireless explaining player covering channels from 64 to 256 and wireless explaining controllers. Each wireless explaining controller comprises a shell and a control circuit arranged in the shell, and the control circuit comprises a power circuit, a radio frequency identification module, a key decoding circuit, a FLASH storage, a CPU, an audio decoding circuit, a D / A conversion unit and an audio amplifier. By means of the wireless explaining player covering the channels from 64 to 256, frequency signals of different channel conversion control signals are sent to exhibition areas or scenic spots of multiple areas, tourists in the corresponding areas receive the frequency signals through the wireless explaining controllers, the control circuits are activated to be connected with the corresponding CPUs, so that audio content and / or video content of the FLASH storages can be called out and completely and continuously played, the tourists can hear explaining information of the scenic spots or the exhibition areas clearly and completely, the adjacent scenic spots or the adjacent exhibition areas do not interfere with each other and the tourists themselves cannot interfere with each other.

The invention discloses a diaphragm preparation method and a sounding device. The preparation method includecomprises the steps of adding a flaky filler into silicone rubber, and uniformly mixing theflaky filler with the silicone rubber; molding the mixed material to obtain a diaphragm. The diaphragm preparation method and the sounding device have the advantages that through modification of the silicone rubber by the flaky filler, the mechanical strength and the damping performance of the diaphragm can be enhanced, and accordingly, the listening performance of the sounding device can be improved.

The invention provides an equipment control method and device. The method is applied to an audio system, the audio system comprises a first audio device and a second audio device, and the method comprises the following steps: respectively using the first audio device and the second audio device to receive wireless signals sent by a terminal; determining the difference between the feature information of the wireless signal received by the first audio device and the feature information of the wireless signal received by the second audio device; when the audio system plays audio, adjusting the phase difference of the audio signals output by the first audio device and the second audio device according to the difference, so that the terminal receives the audio signals output by the first audiodevice and the second audio device at the same time, and a user obtains the optimal listening effect. Through adoption of the method, the user carrying the terminal can obtain the optimal listening effect at different positions, and the user experience is very good.

The invention discloses a volume real-time compensation method, electronic equipment and a readable storage medium, the volume real-time compensation method is applied to an intelligent camera, and the volume real-time compensation method comprises the following steps: detecting whether the position of a to-be-detected target is changed in real time through the intelligent camera; if the position of the to-be-detected target changes, detecting the actual distance between the intelligent camera and the to-be-detected target, and obtaining the current sound pressure level of a sound system in the intelligent camera; and performing volume real-time compensation on a sound system in the intelligent camera according to a volume compensation value determined by the current sound pressure level and the actual distance. The technical problem of poor listening effect of a sound system in an intelligent camera due to user position change in the prior art is solved.