35results about How to "Improve listening quality" patented technology

A control logic means preferably for a receiver comprising a jitter buffer means adapted to receive and buffer incoming frames or packets and to extract data frames from the received packets, a decoder connected to the jitter buffer means adapted to decode the extracted data frames, and a time scaling means connected to the decoder adapted to play out decoded speech frames adaptively. The control logic means comprises knowledge of whether a state recovery function is available and is adapted to retrieve at least one parameter from at least one of the jitter buffer means, the time scaling means, and the decoder, to adaptively control at least one of an initial buffering time of said jitter buffer means, the knowledge of the availability of the state recovery function, and a time scaling amount of said time scaling means from the time scaling means or the decoder.

The invention discloses an earplug cushion for an earphone. The earplug cushion for an earphone comprises a central tube part with a sound outlet and a sound inlet fitting with the sound guide tube of the earphone, an outer shield part extending from the sound outlet to the sound inlet for being inserted into the canal, and a buffer part disposed at the tube part next to the sound outlet; according to special and effective design for the buffer part, when the earphone is worn with the earplug cushion, the buffer part becomes twisted and deformed to be adapted to the canal in different sizes snugly without clearances for decreasing oppression to the canal and enhancing quality of the listened sound.

A very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.

A method, an electronic apparatus and a wireless earphone of choosing a master wireless earphone in a wireless earphone set are provided. The method is adapted to the electronic apparatus pairing with the wireless earphone set. The method includes: transmitting first testing signals to a first wireless earphone in the wireless earphone set, and receiving first acknowledgment signals; transmitting second testing signals to a second wireless earphone in the wireless earphone set, and receiving second acknowledgment signals; determining whether a first ratio of the first acknowledgment signals to the first testing signals is higher than a second ratio of the second acknowledgment signals to the second testing signals; if yes, choosing the first wireless earphone as a master earphone, and setting the second wireless earphone as a slave wireless earphone; if no, choosing the second wireless earphone as the master earphone, and setting the first wireless earphone as the slave wireless earphone.

The embodiment of the invention relates to a control method and a control device. The method comprises the steps of acquiring audio information of current environmental noise, acquiring intensity value of current environmental noise according to current audio information, amplifying or lessening collected local call audio information according to intensity value of current environmental noise, and sending the amplified or lessened local call audio information.

The invention discloses an audio object coding and decoding method based on frequency spectrum shifting. This method proposes a strategy including global shifting and local shifting to reduce aliasing distortion. In the encoding stage, time-frequency transformation is first performed to obtain the spectrum information of multiple input signals; then, it is judged whether aliasing occurs in each subband, and the aliased areas will be sorted according to the degree of aliasing; after determining the time-frequency area that needs to be moved, The aliased time-frequency points are moved to the non-aliasing area, and there are two strategies for moving the time-divided overall shift and local shift. The overall shift can greatly reduce the shift information that needs to be recorded; finally, the downmix signal and side information are synthesized into a code stream. In the decoding stage, the time-frequency components are first restored to their original positions according to the moving information, and then decoded according to the joint audio object coding framework SAOC. The present invention moves aliased time-frequency information to a non-aliased area in a downmixing process by using a spectrum shifting strategy, thereby reducing aliasing distortion and improving decoded audio quality.

The invention discloses an adaptive earplug device, which includes an earplug body matched with the ear canal, and an adaptive sound transmission tube assembly, which is installed on the earplug body; the earplug body includes a shell and a support bar , there is at least one support bar arranged inside the housing, the middle part of the housing is provided with a mounting column, one end of the support bar is connected to the mounting column, and the other end is connected to the inner wall of the housing; the adaptive sound transmission tube assembly includes a sound transmission tube , an adjuster and a sleeve, the sleeve is set on the mounting column, the sound transmission tube is set inside the sleeve through the adjuster, one end of the sleeve extends into the earplug body, and the other end extends out of the ear canal. Also disclosed is a hearing aid device, which includes the above-mentioned earplug device and a host. Earbuds ensure both wearing comfort and improved listening quality. The above-mentioned earplug device is applied to the hearing aid device, and the sound from the speaker in the main unit can be transmitted to the human ear through the sound transmission tube, so that the overall wearing comfort is high and the listening quality is high.

The invention discloses a voice forecasting linkage method and a voice forecasting linkage system based on call requirements. The system comprises an ACD automatic telephone traffic distribution module, an IVR automatic voice answering module, a CTI regulation and control integrated module, en exchange board, a PBX queuing module and a terminal monitoring module; the voice service method can select priority, and timely and effectively answer calls from users according to call requirements of users, thus improving service quality; according to the method, under the premise of reducing operational cost, achieving centralized seat management in the whole province or large areas, and performing centralized operation on calling systems, calls are answered according to belonging areas, and customer service staffs are convenient to communicate; skill groups in cities and counties only need to distribute computers and IP power grids, no gateway and IVR voice answering system are needed, no device redundancy exists, the priority groups are automatically selected according to customer requirements, different priority groups are different in distribution sequences and distribution processes, calls can be effectively and selectively answered, and improvement on answering quality is facilitated.

The invention discloses a voice interaction management method and a voice interaction management system based on telephone traffic forecasting. The voice interaction management system comprises a telephone traffic forecasting module, an ACD automatic telephone traffic distribution module, an IVR automatic voice answering module, a CTI regulation and control integrated module, a PBX queuing module and a terminal monitoring module; the voice service method can select priority, and timely and effectively answer calls from users according to call requirements of users, thus improving service quality; according to the voice interaction management method, under the premise of reducing operational cost, achieving centralized seat management in the whole province or large areas, and performing centralized operation on calling systems, calls are answered according to belonging areas, and customer service staffs are convenient to communicate; skill groups in cities and counties only need to distribute computers and IP power grids, no gateway and IVR voice answering system are needed, no device redundancy exists, the priority groups are automatically selected according to customer requirements, different priority groups are different in distribution sequences and distribution processes, calls can be effectively and selectively answered, and improvement on answering quality is facilitated.

The invention discloses an audio object encoding method suitable for a personalized interactive system. In the encoding stage, the invention first transforms multiple audio objects to be encoded from the time domain into the frequency domain by frame division and windowing; according to each object Sort the energy of each step to determine the coding order of the objects; extract the coding objects and the corresponding downmix signals of each step in a loop, and calculate the parameters and residuals of each step accordingly; use singular value decomposition to decompose and compress the large-sized residual matrix; Combining the final mixed signal, parameter and residual decomposition matrix into a code stream. In the decoding stage, the decomposition matrix is ​​used to reconstruct the residual; then, according to the residual and parameters of each object, the objects are decoded and reconstructed from the downmix signal step by step. The invention can simultaneously ensure low code rate and high quality reconstruction of each audio object through sequential multi-step codec and residual decomposition.

The invention discloses an earphone for communication. The earphone comprises a rubber sleeve, rubber connecting rods are connected to the inner walls of the two sides of the rubber sleeve in a sliding manner; connecting blocks are fixed to the ends, away from the rubber sleeves, of the two rubber connecting rods through screws. A first earphone shell and a second earphone shell are fixed to the outer walls of the opposite sides of the two connecting blocks through screws correspondingly. Grooves are formed in the outer walls of the opposite sides of the first earphone shell and the second earphone shell, loudspeakers are fixed to the inner walls of the grooves through screws, and connecting columns are fixed to the inner walls of the sides, away from the first earphone shell, of the second earphone shell through screws. The fabric is good in air permeability;, Compared with the prior art, the earphone has the advantages that the wearing comfort is obviously improved, the influence onthe normal work of the earphone due to the fact that sweat permeates into the earphone shell in the wearing process is avoided, the external sound interference is isolated, the listening quality of the earphone is improved, components in the earphone are prevented from being burnt out after long-time working, and the service life of the earphone is obviously prolonged.

The invention discloses a method and a system for regulating and controlling voice service based on call requirements. The system comprises an ACD automatic telephone traffic distribution module, an IVR automatic voice answering module, a CTI regulation and control integrated module, an exchange board, a PBX queuing module and a terminal monitoring module; the voice service method can select priority, and timely and effectively answer calls from users according to call requirements of users, thus improving service quality; according to the method, under the premise of reducing operational cost, achieving centralized seat management in the whole province or large areas, and performing centralized operation on calling systems, calls are answered according to belonging areas, and customer service staffs are convenient to communicate; skill groups in cities and counties only need to distribute computers and IP power grids, no gateway and IVR voice answering system are needed, no device redundancy exists, the priority groups are automatically selected according to customer requirements, different priority groups are different in distribution sequences and distribution processes, calls can be effectively and selectively answered, and improvement on answering quality is facilitated.