227results about How to "Guaranteed sound quality" patented technology

The invention discloses an automatic bone conduction playback mode switching method and device and an intelligent watch. The method includes the steps that the pressure, generated by making contact with the human bone, of a bone conduction device preset on a terminal is detected; whether the pressure reaches the set threshold value or not is determined, the bone conduction device is started, and the terminal is switched to the bone conduction playback mode. By the adoption of the technical scheme, when the wearable intelligent terminal makes contact with the skull or the face bone, the bone conduction playback mode can be switched into in an self-adaptive mode according o the pressure, the playback sound quality of the wearable intelligent terminal is guaranteed, the privacy of a user can be guaranteed by starting the bone conduction playback mode at proper time, and user experience is improved.

The invention relates to the field of voice technologies, in particular to a self-adaptive denoising method based on sub-band noise analysis. The method includes the steps that firstly, framing and short time frequency domain transformation are conducted on input time domain audio signals with noise, and then frequency domain audio signals with noise are generated; secondly, a noise energy spectrum of the frequency domain audio signals with noise is estimated through a minimum value tracking method; thirdly, the posterior signal to noise ratio and the prior signal to noise ratio of the noise energy spectrum are calculated; fourthly, through a nonlinear gain extension method, denoising gains of all time frequency units are calculated through the posterior signal to noise ratio and the prior signal to noise ratio; fifthly, smoothing filtering is conducted on the denoising gains of all the time frequency units to reduce tone quality distortion; sixthly, the denoising gains act on all the time frequency units of the audio signals with noise in the first step, and then denoised frequency domain audio signals are acquired; seventhly, short time frequency domain inverse transformation is conducted, and then the final denoised time frequency audio signals are acquired and output. According to the method and system, stable noise in target signals can be greatly lowered.

The invention relates to a modified material film of a micropore film composite nonwoven fabric for waterproof dustproof acoustically transparent expanded polytetrafluoroethylene (PTFE) and a preparation method thereof. The modified material film comprises a micropore film layer of an expanded PTFE, one or more layers of nonwoven fabrics compounded and connected with the micropore film layer surface of the expanded PTFE; the modified material film can protect a loud speaker, a microphone, an alarm and the like while keeping the tonal quality of an apparatus, further prevents common liquids such as beverage, beer, coffee and the like, and can tolerate the water soaking; the protection level can achieve IP67; the dewing phenomenon in the loud speaker, the microphone and the alarm can be eliminated; the capability of tolerating the environment temperature change is improved, the shell design difficulty and the manufacturing cost are reduced, the cold flow property of the micropore film of the expanded PTFE is solved, and the processability is improved; the modified material film can be manufactured as dustproof waterproof venting acoustically transparent bolts, pasters and other ready-to-use components with a series of standard venting product sizes for the loud speaker, the microphone, the alarm and the like; and therefore, the design application is convenient.

The invention provides a coding method and a decoding method for voice data. The coding method comprises the following steps: acquiring an original audio frequency, obtaining voice segment data by eliminating non-voice data in the original audio frequency through end point detection; extracting a spectrum parameter, a fundamental sound period, and other parameters for each frame of the voice data, calculating a gain mean of continuous multi-frame voice data, conducting vector quantization for the spectrum parameter, and nonlinearity quantification for the fundamental sound period and the gain mean through a vector code book; and conducting coding for various quantified voice parameters to generate a voice data package. The decoding method comprises the following steps: conducting decoding for a received voice data package; extracting a spectrum parameter, the fundamental sound period, the gain mean, and other parameters; conducting forecasting for an excitation parameter and an energy changing track through the parameters; and finally, synthetizing voice through a vocoder. The method can keep a higher voice quality at extremely low coding rate.

The invention discloses a nonlinear echo suppression method. The nonlinear echo suppression method comprises the following steps: performing Fourier transform on a far-end reference signal x, a microphone received signal d, a linear estimated echo signal y and a residual signal e to obtain corresponding frequency domain signals Xf, df, yf, and ef; performing correlation coefficient calculation onthe xf, df, yf, and ef; smoothing the above correlation coefficients; and averaging the smoothed correlation coefficients for a frequency point of a nonlinear echo-easy product; constructing a nonlinear echo suppression function by using the correlation coefficient and the mean value; eliminating the nonlinear echo through the nonlinear echo suppression function; and performing Fourier transform on a signal where the nonlinear echo is eliminated, and obtaining a time domain output signal. The method effectively improves the echo suppression ratio of the system under the condition of ensuring the sound quality of the output signal, reduces the distortion of the near-end speech signal, and has a small calculation amount, and has good application prospect in a voice interaction system.

The invention is suitable for communication fields and provides a method, a system and an IAD (Integrated Access Device) for processing a voice message. The method comprises the following steps of: intercepting a message voice pack of a remote user in a PSTN (Public Switched Telephone Network) network or an IP (Internet Protocol) network by the IAD; extracting voice data in the voice pack by the IAD; and storing the voice data into a local voice mail database by the IAD after preprocessing the voice data. According to the method, the system and the IAD for processing the voice message, through a manner that a voice mail is arranged in the IAD, a method for accessing the voice mail by three manners including a remote phone, a local phone and a Web page is realized, the requirements of different users on voice mail services on various occasions can be met, and the dependence of a pure software voice mail on a PC (Personal Computer) machine can be removed.

The invention relates to a sound box module and especially relates to a loudspeaker and a manufacturing method thereof. Sound is generated by the loudspeaker in a manner of generating mechanical vibration under an action of an electromagnetic force. The loudspeaker comprises a sound box module which comprises a horn, a driven radiator and a sound box panel, wherein a mounting hole is formed on the sound box panel; the horn is arranged at the mounting hole and is integrally formed with the sound box panel in the manner of embedding ejection; a secondary mounting hole is formed on the sound box panel on one side of the mounting hole; and the driven radiator is arranged in the secondary mounting hole and is integrally formed with the sound box panel in the manner of embedding ejection. The loudspeaker has the advantages of small volume, small thickness, simple structure and enhanced low pitch. The loudspeaker can be used as a built-in loudspeaker or an external loudspeaker of a modern panel personal computer or intelligent mobile phone, and the like. In order to ensure an excellent sound effect of the sound box, an embedding ejection integrating manufacturing process is adopted, thereby maximally ensuring the sound effect of the sound box, greatly shortening the working time and promoting the production efficiency.

The invention discloses a dual-positioning type bone conduction loudspeaker device. The device comprises a supporting part; a magnetic assembly which at least comprises one magnetic part; and a positioning assembly which at least comprises two elastic elements. The magnetic assembly and the positioning assembly are disposed on the supporting part. The two elastic elements are respectively connected with the magnetic assembly. The magnetic assembly and the supporting part remain relatively stationary. Through the arrangement of a dual-positioning structure, the device guarantees that the relative position of a magnetic gap and a voice coil remains the initial position when a bone conduction loudspeaker works, thereby preventing noise sound and guaranteeing the sound quality of the loudspeaker. Moreover, the device can be thinner, is smaller in size, and is lighter in weight. The device improves the sensitivity while reducing the noise sound, and reduces the power consumption.

The sound output system comprises a sound output device and an audio output device. The corresponding method comprises: detecting whether the sound output device is inserted into the audio output device; if yes, using the sound output device to send stored compensation feature parameter to another device; adjusting audio signal, sending signal to the sound output device; using latter to convert the signal into analog audio signal for output. This invention can ensure the effect of audio playback.

The invention discloses a method and a device for adaptive discontinuous voice transmission. The method includes: during adaptive discontinuous voice transmission, determining whether to transmit a silence insertion descriptor or not according to a current voice signal frame and spectral information of a previous silence insertion descriptor. By the method and device, the problems that flexibly monitoring signal change by means of fixed intervals fails in the prior art and necessity of computation on multiple parameters such as linear prediction for the use of the means of variable intervals causes high computation complexity can be solved. Transmission is directly performed frequency domains by the method and device, signal change can be well tracked, and acoustic fidelity is guaranteed while low average bitrate is kept.

The invention discloses a sound box. The sound box comprises a box body and a loudspeaker arranged on the box body. The box body comprises an outer box and an inner box arranged in the outer box, the inner box is made of wood material, and the outer box is made of plastic. Through the inner box made of the wood material and the outer box made of the plastic, the tone quality of the sound box can be ensured, and the appearance of the sound box can be manufactured into the required shape.

The invention discloses a mobile terminal and a sound outlet switching method. The mobile terminal comprises a frame body and an opening for the frame body to extend or retract. The frame body is internally provided with a sound cavity, a telephone receiver, an optical device module, and a first sound outlet and a second sound outlet which are communicated with the sound cavity, and the telephonereceiver is communicated with the sound cavity. The mobile terminal further comprises a first sound guide channel, a second sound guide channel and a telescopic mechanism. Under the driving of the telescopic mechanism, the optical device module can at least partially extend out of the opening or completely retract into the mobile terminal through the opening, so that one sound outlet is communicated with the corresponding sound guide channel, and other sound outlets are separated, and the problem that the thickness of the mobile terminal is thickened or the length of the mobile terminal is toolong due to the fact that a plurality of receivers are arranged does not need to be considered. Meanwhile, the problems of poor tone quality, small volume and the like caused by simultaneous sound output of other sound outlets are effectively avoided, and the tone quality and the volume of the sound transmitted by the sound outlets can be guaranteed under the condition that the stacking space ofthe mobile terminal is allowed.

The invention discloses a method, a device and a system for outputting HiFi (High Fidelity) audio frequency. The method comprises the following steps that after receiving PCM (Pulse Code Modulation) code stream and PCM format information, an audio frequency manager Audio Flinger of a mobile terminal transmits the PCM code stream and the PCM format information to a virtual HiFi sound card preconfigured in the mobile terminal in a unvarnished manner; the virtual HiFid sound card transmits the PCM code stream and the PCM format information to a Mobile-HiFi convertor by virtue of a communication linkage established between the virtual HiFi sound card and the Mobile-HiFi convertor; and the Mobile-HiFi convertor generates an I2S signal based on the received PCM code stream and PCM formation information and outputs the I2S signal to an audio play device. According to the technical scheme adopted by the invention, the mobile terminal can be converted into a wireless digital turntable, so that when the mobile terminal serves as a sound source, the HiFi output of sound quality is ensured, and problems existing in the prior art are well solved.

The invention provides a method for multi-microphone sound mixing of a video conference system. A maximum echo channel is decided and excluded through a delay time between sounds acquired by various microphone channels and loudspeaker playing sounds and energy capacity of the acquired sounds, and then a sound acquired by an optimum channel is selected as a sound mixing source and a sound mixing weight of the selected channel is determined based on frame energy values acquired by various optional channels and correlation of the acquired frame energy values, and finally, sound mixing output is performed according to the selected channel and the determined sound mixing weight. By the aid of the method, the sound acquired by the optimum channel is selected as an optimum sound mixing source, and all algorithms are designed bases on self-adaption, so the optimum sound mixing source can be adjust dynamically, and acquisition range and tone quality of the sound are guaranteed.

The invention relates to the field of electronic equipment, and provides an earphone interface circuit and electronic equipment. The earphone interruption detection end of the earphone interface circuit is connected to a right sound channel pin or a left sound channel pin of an interface; and the earphone interface circuit also comprises an earphone plug-in detection circuit which has the same level as the earphone interruption detection end, clamps the earphone interruption detection end, and is connected to the right sound channel pin or the left sound channel pin. Because the earphone plug-in detection circuit is arranged, earphone detection delay is reduced, and the quality and volume of sound are ensured simultaneously; and the earphone interface circuit is used in the electronic equipment, and a good using effect of the electronic equipment can be ensured.

An audio switching system of a mobile communication terminal comprises a central processing unit (CPU), an audio amplifier, a headset, a loudspeaker, a power management unit (PMU) and a switch module. The CPU is respectively electrically connected to the control switch module, the PMU and the audio amplifier and outputs left-track audio signals and right-track audio signals. The switch module selectively transmits the right-track audio signals to the headset. The PMU selectively transmits the left-track audio signals to the headset or the audio amplifier. The audio amplifier amplifies the left-track audio signals and transmits the amplified left-track audio signals to the loudspeaker. When the headset is used, the switch module and the PMU are respectively communicated with the CPU and the headset so as to transmits the left-track audio signals and the right-track audio signals to the headset, and the switch module disconnects the CPU and the loudspeaker at the seam time; when the loudspeaker is used, the CPU turns on the audio amplifier, the PMU is communicated with the CPU and the audio amplifier, the audio amplifier amplifies the single-track audio signals and transmits the amplified audio signals to the loudspeaker, and the switch module disconnects the CPU and the headset at the same time.

The invention discloses a voice processing method and device. The method is applied to a terminal having two microphones on the top, the two microphones are located on the front and the back of the terminal respectively, and the method is applied to a non-video call scene. The method comprises acquiring a voice signal by using the two microphones when it is detected that the camera of the terminalis in a capture state; calculating a sound pressure difference between the two microphones according to a first predetermined algorithm based on the acquired voice signal; determining whether the sound pressure difference satisfies a sound source direction determination condition; if so, determining whether a backward voice signal is included in the voice signal according to the sound pressure difference, wherein the backward voice signal is a voice signal whose sound source is located behind the camera; if it is determined that the voice signal includes the backward voice signal, filtering out the backward voice signal in the voice signal. In this way, the method locates the sound source based on the sound pressure difference in a low SNR scene and can improve the pickup accuracy of thesound source within an imaging range.

The invention discloses a permanent magnet for a loudspeaker and a processing technology thereof. The permanent magnet comprises the following components by weight percentage: 10-14.5 percent of neodymium, 1.5-2.4 percent of calcium carbonate, 2.2-2.6 percent of titanium, 8-12.5 percent of zinc oxide, 2-4.5 percent of boron, 3-4.5 percent of aluminum oxide, 0.2-0.8 percent of gallium and balance of iron oxide. The raw materials are treated through oxidization, air-stream milling, molding, isostatic pressing, oil peeling, sintering and post-processing to obtain products. By using the iron oxide as the foundation bed of the permanent magnet, when the molar ratio of ferric iron to ferrous iron in the iron oxide is 2:1, the maximum magnetic energy product of the permanent magnet is the highest, the magnetic induction coercivity is high, the connection of grains in the permanent magnet is stable after long-time use, the sound quality of the loudspeaker is guaranteed, the long-time use of the loudspeaker is facilitated, different alloy metals can be added according to loudspeakers at different frequencies, the application scope is wide and the overall performance of the loudspeaker is improved.

A cross-voice eliminative installment and the method, use for eliminate the procreant cross-voice by the duplex soundtrack frequency two speakers broadcast that includes 3D information: use the temper unit and filter to do the delay processing and the filter processing respectively for the signal list of the left soundtrack and right soundtrack, get the left soundtrack delayed signal list, the right soundtrack delayed signal list, the left soundtrack filtered signal list and the right soundtrack filtered signal list; the left soundtrack delayed signal list and the right soundtrack filtered signal list add by the adder, then output from the left speaker, simultaneity the left soundtrack filtered signal list and the right soundtrack delayed signal list add by the adder, then output from the right speaker.

The invention discloses a method for processing audio information and an electronic device. The method includes the steps of obtaining at least one hardware parameter of an audio output unit; based on the at least one hardware parameter, determining a set of audio playing parameters according to a preset rule, wherein the audio playing parameters are matched with the hardware parameter of the audio output unit; obtaining a first instruction, and determining a first multimedia file according to the first instruction; controlling the audio output unit to play the first multimedia file according to the audio playing parameters. By means of the method and the electronic device, it can be guaranteed that the audio output unit can be kept in the optimal playing state.

The invention discloses an earplug matched with the ear in shape and relates to the technical field of acoustic products in contact with the ear. The earplug matched with the ear in shape comprises an earplug front-cavity shell and an earplug rear-cavity shell, wherein the earplug front-cavity shell and the earplug rear-cavity shell are combined together. A trumpet single body is contained in a space defined by the earplug front-cavity shell and the earplug rear-cavity shell. A sound outlet is formed in the earplug front-cavity shell. An ear coupler covers the outer portion of the earplug front-cavity shell. A through hole used for communicating the sound outlet and the outside environment of the earplug is formed in the ear coupler. The ear coupler is made of foam or soft silica gel or a memory sponge. The earplug matched with the ear in shape solves the technical problems that an in-ear type earplug is not matched with the ear and not comfortable in wearing in the prior art. The shape of the earplug matched with that of the ear can change along with the shape of the ear, so that a user does not feel uncomfortable or painful when the earplug is worn by a user for a long time.

The invention discloses an audio device and a spatial noise reduction method thereof. The audio device comprises a microphone array and a horn array. The microphone array comprises a plurality of first microphones arranged in different spatial directions. The horn array comprises a plurality of first horns arranged one by one corresponding to the first microphone. The method includes acquiring a first sound signal received by each of the first microphones; the first sound signal includes a user sound signal and a noise sound signal. Controlling a corresponding first horn to play a suppressed sound signal for the noise sound signal according to the preset sound information and each first sound signal; the suppressed sound signal includes an inverse sound wave for suppressing a correspondingnoise sound signal, and the preset sound information includes user sound information acquired in advance. The invention enlarges the suppression range of the spatial noise in the use process, enhances the noise reduction effect, and also ensures that the sound signal of the user is not suppressed, so as to be convenient for the user to use.

The invention relates to a multi-point excitation loudspeaker array which is characterized in that: the array comprise a basin rack which is provided with at least five independent sound cavities, a number of the sound cavities is an odd number, each sound cavity is provided with a loudspeaker unit, all loudspeaker units are arranged in straight lines in the basin rack and are on a same installation plane, and distances between adjacent loudspeaker units are same. The invention has the characteristics of ingenious structure, simplicity and compactness, the loudspeaker units are arranged in an easily realized mode, when adding the loudspeaker units, small directive change of the whole array is ensured, total harmonic distortion and unevenness of the array are decreased, thus a sound pressure level of the array is raised, sound quality and reliability are ensured, and the array can be designed as a component integrated in a flat television or an external accessory of the flat television to improve the sound quality further. The multi-point excitation loudspeaker array has the advantages of good frequency response, excellent directive, large bearing power, small distortion, a high sound pressure level and the like.

The invention relates to a push-pull loudspeaker. The loudspeaker comprises a magnetic circuit structure, a connecting base, a first voice coil, a first frame, a second voice coil and a second frame. According to the push-pull loudspeaker, the first voice coil and the second voice coil are arranged in magnetic gaps of the magnetic circuit structure, and the first voice coil and the second voice coil are fixedly connected. Under the action of a magnetic field, the first voice coil and the second voice coil move together to drive a first vibrating diaphragm and a second vibrating diaphragm which are installed in an opposite mode to move together, so that the effect that one of the two vibrating diaphragms is pushed and the other is pulled is achieved, vibrating is more balanced, and low-frequency distortion is reduced; meanwhile, the first vibrating diaphragm, the connecting base and the second vibrating diaphragm form a resonant cavity body, medium-high frequency response components are filtered out, and the low-frequency tone quality is guaranteed.

The embodiment of the invention provides a sound quality optimization method, a feedback noise reduction system, an earphone and a storage medium. The method comprises the steps that a mixed signal acquired by a microphone in the feedback noise reduction system at present and a useful signal to be sent into a loudspeaker in the feedback noise reduction system at present are acquired; an environment noise signal contained in the mixed signal is acquired according to the mixed signal and a useful signal transmitted to the microphone at present; the environment noise signal contained in the mixedsignal is output to a noise reduction filter in the feedback noise reduction system to allow the noise reduction filter to generate a noise reduction signal corresponding to the environment noise signal; and the noise reduction signal and the useful signal to be sent to the loudspeaker at present are sent to the loudspeaker together to obtain a useful signal which has the lossless sound quality.According to the method, the influences of the feedback noise reduction process on the useful signal can be effectively improved, and the sound quality of the useful signal is guaranteed.

The invention relates to the audio signal processing technology and discloses a method and equipment for processing digital audio in variable speed. In the invention, a pair of perfect reconstructing window functions WL and WR with amplitude attenuation and increase characteristics is used to act on original digital audio according to different delays to obtain a pair of windowing data; and an audio waveform is reconstructed by using the windowing data to obtain audio after variable-speed process. The defection on a fundamental tone period and the relativity of the audio and the time-frequency conversion are avoided, so that the calculation amount is extremely small. In addition, the playing time of the contents is prolonged or shortened by using the compaction and the introduction of theself waveform of an audio signal without changing the audio waveform, so that the original tone quality can be better maintained.

The invention discloses a sound signal processing apparatus and a sound signal processing method. The sound signal processing apparatus includes a sound source direction determination unit and a filter processing unit. The sound source direction determination unit determines sound source directions with respect to sound signals of a plurality of channels for respective first to n-th bands. The filter processing unit includes first to n-th filters which are connected in series and configured to boost or attenuate the sound signals with respect to the first to n-th bands. The respective first to n-th filters perform boosting or attenuation based on the sound source directions of the first to n-th bands which are determined by the sound source direction determination unit.

The invention provides an audio coding method and device. The audio encoding method comprises the following steps: acquiring first audio data; obtaining a target code rate and a Bluetooth packet type,wherein the target code rate and the Bluetooth packet type correspond to the current Bluetooth channel condition; obtaining one or more of a bit pool parameter set, a psychoacoustic parameter set anda spectral bandwidth parameter set through a pre-trained neural network according to the first audio data, the target code rate and the Bluetooth packet type; and encoding the first audio data according to one or more of the bit pool parameter set, the psychoacoustic parameter set and the spectral bandwidth parameter set to obtain a to-be-sent code stream. According to the invention, the Bluetooth channel condition can be adaptively matched, and continuous audio hearing experience is brought while the tone quality is ensured to the maximum extent.

The invention provides a wireless earphone playing tone quality prompting method, a mobile terminal and a storage medium, and is applied to the mobile terminal, and the method comprises the steps: obtaining the signal intensity of a wireless signal received by a wireless earphone and a data transmission rate of a currently played audio in real time, inputting the signal intensity and the data transmission rate into a pre-constructed packet loss rate database, screening to obtain the current packet loss rate of the corresponding currently played audio, inputting the current packet loss rate into a pre-constructed tone quality database, screening to obtain corresponding tone quality information, judging whether the tone quality information triggers a prompt function, and if the prompt function is triggered, outputting first prompt information to the wireless earphone. According to the invention, the sound quality information of the audio played by the wireless earphone can be obtained inreal time according to the pre-constructed sound quality database, and the first prompt information is output to the wireless earphone after the sound quality information triggers the prompt function, thereby reminding a user that the user deviates from the high-sound-quality region of the wireless earphone, so that the user can make a response in time, and the user experience is improved.

The invention discloses a method and a system for sound frequency redirection in a virtualization desktop. The method comprises steps of obtaining first audio data through an RDP protocol, transmitting first audio data to a service terminal through a virtual device interface, and coding first audio data hardware code into second audio data by a service terminal. Through above arrangement, the invention can greatly reduce the CPU utilization rate of the client terminal on the premise that the sound quality is guaranteed so as to reduce the whole load of the thin client terminal and break the limitation of the network transmission bandwidth.