79results about How to "Improve voice call quality" patented technology

The invention discloses a determination method and device for a voice communication network. The method comprises the steps that a terminal in support of a VoLTE (Voice over Long Term Evolution) obtains a network frequency point at which the terminal currently resides, and the voice indicator information and signal quality information of the network frequency point, when a voice communication demand is detected, wherein the signal quality information may comprise reference signal receiving power and a signal to noise ratio, and the voice indicator information may comprise a call success rate, a call drop rate and an SRVCC (Single Radio Voice Call Continuity) switching rate; and the terminal determines the voice communication network of the terminal based on the voice indicator information and the signal quality information. In this way, the terminal can carry out comprehensive judgment according to the voice indicator information and signal quality information corresponding to the frequency point of a service cell of the terminal, the suitable voice communication network is selected, the voice call quality of the terminal is improved, moreover, SRVCC switching signal processes are reduced, and network loads are reduced.

The method comprises: when receiving end incorrectly receives current voice data frame, it decide if the previous voice data frame of current voice data frame and the next voice data frame of the previous voice data frame are active voice signals; according to the decision result, makes relevant operation to respectively generate voice data; making composition process for the generated previous voice data frame and the voice data of next voice data frame to generate a voice compensating data frame.

An embodiment of the invention provides a voice enhancement processing method and device. The method includes: decoding bit streams, and obtaining coding parameters of a current voice subframe to be processed, the coding parameters including first generation digital book gains and first self-adaptive code book gains; adjusting the first generation digital book gains, and obtaining second generation digital book gains; determining second self-adaptive code book gains according to the first self-adaptive code book gains and the second generation digital book gains; and adopting quantization indexes of the second generation digital book gains and the second self-adaptive code book gains to replace bits corresponding to the first generation digital book gains and the first self-adaptive code book gains. By adoption of the technical scheme of the invention, an effect of eliminating noise can be effectively improved, and voice communication quality can be improved.

The invention discloses a method and a device for compensating a drop frame after a start frame of voiced sound and guarantees against delaying of compensation to the drop frame after the start frame of the voiced sound. The method includes selecting different manners to deduce pitch delay of a first drop frame following the start frame of the voiced sound on the condition of stability of the start frame of the voiced sound; deducing self-adaptive codebook gain of the first drop frame according to self-adaptive codebooks of one or multiple sub-frames received before the first drop frame, or deducing the self-adaptive codebook gain of the first drop frame according to energy change of time-domain voice signals of the start frame of the voiced sound; and compensating the first drop frame by the pitch delay and the self-adaptive codebook gain deduced. After compensation, each sub-frame of the frame correctly received after the start frame the voiced sound is decoded to acquire the self-adaptive codebook gain, the self-adaptive codebook gain times a scale factor to obtain the new self-adaptive codebook gain of the corresponding sub-frame, and the new self-adaptive codebook gain substitutes for the self-adaptive codebook gain acquired by decoding to participate in voice synthesis. Therefore, error transmission caused by the drop frame can be decreased, and energy for voice synthesis can be controlled.

An apparatus and method for generating a three dimensional (3D) stereo sound signal from a received audio signal in a mobile communication system are provided. In the 3D stereo sound generating apparatus, a low-frequency signal extraction portion extracts a low-frequency signal from a received audio signal, a spatiality generator generates a spatiality signal from the received audio signal, an output mode selector receives the spatiality signal and the low-frequency signal and selects an output mode for a 3D stereo sound signal, and an output portion outputs the 3D stereo sound signal to a predetermined output device according to the selected output mode.

Technical problems to be solved in the invention are that mobile phone is used in a noise environment, which may effectively filter noise, echo and other background noises of non-conversation angle. A method to improve voice call quality of mobile phone is that after the mobile phone is connected, a DSP samples and compares two paths of microphone signals input; the DSP discovers that amplitude difference of the two paths of microphone signals is appropriate and identifies the signals as valid signals and processes the signals relatively; when voice from non-conversation angle is delivered to the microphone, via pickup of the microphone, the DSP discovers that amplitude difference of the two paths of microphone signals exceeds a certain value and views the signal as non-conversation noise and filters the signal. Active effect of the invention lies in that the DSP realizes small angel pickup effect via digital process of two paths of monodirectional and omnidirectional microphones, which filters noise of non-conversation angle and improves voice call quality of mobile phone.

The invention provides a switching method and device of a phone-communication MIC and a mobile terminal, and relates to the field of mobile communication. The method comprises the steps that if the mobile terminal is in a phone-communication state, whether a first MIC serving as the phone-communication MIC is shielded or not is detected; if so, a second MIC is started to serve as the phone-communication MIC. According to the switching method and device of the phone-communication MIC and the mobile terminal, the phone-communication MIC can be timely switched when the phone-communication MIC ofthe mobile terminal is shielded, the voice phone-communication quality is improved, and then the user experience of the mobile terminal is improved.

The invention provides a system in which an analog telephone is bound with a software terminal to realize syncretic communication. The system at least comprises a software terminal, a common analog telephone and a server communicated with the software terminal and connected with the common analog telephone; wherein, the common analog telephone is connected to the server through a common telephone wire, the software terminal is installed on a PC and connected to the server through the TCP/IP network; the software terminal at least comprises an IM processing module, a short message processing module, a voice processing module, a bound telephone processing module, a united account processing module and an address book processing module. Accordingly, by the binding of the analog telephone and the software terminal, the user can not only obtain higher voice communication quality of the common analog telephone but also enjoy the convenience of the intelligent software telephone, and the user can conveniently select the common telephone or the software terminal to answer calls.

The invention discloses a single-channel voice enhancement method and device, a storage medium and a terminal. The method comprises the steps of obtaining a frequency domain amplitude spectrum of a current frame signal based on a received input signal; performing VAD processing on the full band of the current frame signal based on the frequency domain amplitude spectrum of the current frame signalto obtain an initial full band amplitude spectrum gain function of the current frame signal; dividing the whole band into a plurality of sub-bands; performing VAD processing on the plurality of sub-bands of the current signal frame based on the frequency domain amplitude spectrum of the current frame signal and the initial full-band amplitude spectrum gain function, and updating the initial full-band amplitude spectrum gain function according to the VAD processing result of each sub-band to obtain an updated full-band amplitude spectrum gain function of the current frame signal; and calculating according to the frequency domain amplitude spectrum of the current frame signal and the updated full-band amplitude spectrum gain function to obtain a frequency spectrum after speech enhancement.Through the scheme of the invention, non-stationary noise can be effectively suppressed, the voice quality is protected from being lost, and the voice call quality of mobile equipment such as a mobilephone is favorably improved.

An embodiment of the invention discloses a method, a device and a system for transmitting sound package. The method for transmitting the sound package comprises the following procedures: receiving the sound package; caching the received sound package according to the identification of the received sound package; and transmitting the sound package in the cache to a core network a transmitting time which uses a sound package transmitting time period requested by the core network as a period is arrived. The technical scheme of embodiment of the invention can be applied to ascending service of CS over HSPA. Interference eliminating to the ascending sound package is realized through caching the sound package to guarantee that the sound package is transmitted to the core network through a fixed time interval requested by the core network thereby guaranteeing the continuity of sound and increasing the communication quality of sound.

The invention discloses a ZigBee technology-based wireless full-duplex real-time voice transmission device and method. The device is characterized by comprising a device A and a device B, which are same and on an equal footing, wherein the device A, as well as the device B, adopts the structure that voice signals are converted into analog voltage signals through an electret microphone; voltage signals are converted into the voice signals through an electrodynamic loudspeaker; a voice processing circuit converts the analog voltage signals output by the electret microphone into digital signals which are then sent to an embedded microcontroller circuit, and, at the same time, also receives and converts the digital voice signals output by the embedded microcontroller circuit into the analog signals which are then sent to the electrodynamic loudspeaker; and the embedded microcontroller circuit receives the digital voice signals output by the voice processing circuit, so as to compress the signals in a 'ping-pang' manner and then send the compressed signals to a ZigBee RF circuit for RF transmission, and also receives the data output by the ZigBee RF circuit, so as to decompress the data and then send the decompressed data to the voice processing circuit in a in a 'ping-pang' manner. The method and the device provided by the invention have the advantages of simple structure, and lower consumption and cost.

The embodiment of the invention relates to a voiceprint recognition technology, and discloses a voice call noise elimination method, which comprises the following steps of: carrying out voice endpointdetection on a call audio to obtain a human voice set; performing voice feature extraction on the human voice set to obtain a voice feature set; intercepting a to-be-detected voice feature set of which the cumulative duration is a preset duration threshold from the voice feature set according to a time sequence to obtain a plurality of to-be-detected voice feature sets, performing clustering processing on each to-be-detected voice feature set, and scoring the clusters; and according to the score, dividing the human voice set into a first speaker voice and a second speaker voice, distinguishing a background human voice from the first speaker voice and the second speaker voice, and deleting the background human voice from the human voice set. The invention also relates to a blockchain technology, and the call audio can be stored in the blockchain. The background voice in the voice call can be deleted, so that the voice call quality is improved.

Embodiments of the invention provide a method and an apparatus for canceling co-channel interference. The method comprises the steps of selecting candidate co-channel interference cells from a plurality of potential co-channel interference cells according to business measurement implemented in time slot for carrying out downlink receiver; and determining a target candidate co-channel interference cell from the candidate co-channel interference cells based on measurement results directed to the candidate co-channel interference cells, so as to eliminate the co-channel interference from the target co-channel interference cell. The embodiments of the invention can rapidly and accurately determine the co-channel interference cells and efficiently realize elimination of the co-channel interference.

The embodiment of the invention provides a filtering method and device and electronic equipment, which are applied to the technical field of communication and are used for solving the problem of poorvoice call quality of the electronic equipment. The method comprises the following steps: acquiring an estimated echo signal corresponding to a first voice signal output by a loudspeaker of the electronic equipment; subtracting the estimated echo signal from a first audio signal received by a microphone of the electronic equipment to generate a second audio signal, with the first audio signal comprising a second voice signal and a first echo signal fed back to the microphone by a loudspeaker; calculating a weighted filtering parameter according to an energy parameter of a residual echo signalin the second audio signal, with the residual echo signal being a residual echo signal in the second audio signal; and filtering the second audio signal according to the weighted filtering parameter to generate a target audio signal. The application is applied to an acoustic echo elimination scene.

The present invention is to provide a building intercom alarm system which is characterized in that the building intercom alarm system comprises intercom devices, alarm buttons, alarm input wires, processors, first decoders, microphones, loudspeakers, cameras, indoor units, floor distributors, connection network cables, a network switch, a center management main machine, a monitoring server, a second decoder, an alarm device, a visual intercom and doorway management machines. Each indoor unit is internally provided with the intercom device; the lower end of each intercom device is provided with the processor; the intercom device is electrically connected with the processor; the lower end of each processor is provided with the first decoder; and the processors are electrically connected with the first decoders respectively. The beneficial effects of the building intercom alarm system are that the system can be in wired or wireless communication with a residence community property management center or community guards through the center management main machine, thereby achieving the safety protection functions of preventing thefts, disasters and coal gas leakage and the like, and providing maximum guarantee for life and property safety of a house owner.

The invention provides a method for circuit switched fallback, a CSCF (Call Session Control Function) server, a mobile terminal, a mobility management entity and a system for circuit switched fallback. The method comprises the following steps of receiving protocol signaling for initiating VoLTE (Voice over LTE) voice service from the mobile terminal; determining whether to send a service query command to the mobility management entity of the mobile terminal according to the protocol signaling; and receiving a query result from the mobility management entity and sending a circuit switched fallback command to the mobile terminal according to the query result in order to make the mobile terminal execute a circuit switched fallback operation. Through the technical scheme of the invention, the mobile terminal can be effectively guided to execute the circuit switched fallback operation to guarantee continuity of a voice call, so that the voice call quality of the mobile terminal supporting IMS (IP Multimedia Subsystem) service is effectively improved, the user experience is promoted, and meanwhile, an LTE (Long Term Evolution) network can select the voice communication mode of the mobile terminal according to the current network state.

The invention discloses a network service processing method and device, and belongs to the technical field of communication. The method comprises the steps of determining whether a received real-time transport protocol (RTP) packet is abnormal or not under the condition that a first voice call service is performed based on a first network; and when it is determined that the received RTP packet is abnormal and the signal quality parameter of a second network is higher than a preset threshold, switching to the second network to perform a second voice call service, the first network being one of a long term evolution (LTE) network and a wireless communication (WIFI) network, and the second network being the other of the LTE network and the WIFI network.

The invention relates to a voice enhancement method and device, and the method comprises the steps: calculating the voice existence probability of a current frame of audio signal, wherein the voice existence probability represents the existence probability of a voice signal in the current frame of audio signal; utilizing the voice existence probability to obtain a noise variance of a next frame ofaudio signal; and performing voice enhancement on the next frame of audio signal by using the noise variance of the next frame of audio signal. Voice enhancement is performed on the next frame of audio signal based on the voice existence probability so that the noise suppression level can be effectively enhanced and the loss of the voice signal can be reduced, the integrity and intelligibility ofthe voice signal after noise suppression can be guaranteed and the voice call quality can be enhanced.

The invention relates to a paging method and apparatus, and a computer readable storage medium for solving the technical problem that if an existing communication device initiates paging in the case that the currently connected network state is not good, the possibility of call drop is relatively large. The paging method comprises the following steps: when a communication device generates a pagingrelated event, querying a historical call drop record of a base station cell currently accessed by the communication device; when the historical call drop record meets a network switching condition,switching the communication device to a 2G/3G network; and initiating paging in the 2G/3G network according to a paging request.

Provided are a wireless communication method, terminal equipment and network equipment, the terminal equipment can trigger and release at least one wireless connection other than a first wireless connection among a plurality of wireless connections maintained by the terminal equipment, therefore, the user experience can be improved when adverse effects such as terminal body temperature rise, powerconsumption increase and terminal battery power shortage occur. The wireless communication method comprises the following steps: the terminal equipment sends first information, wherein the first information is used for requesting to release at least one wireless connection except a first wireless connection in a plurality of wireless connections maintained by the terminal equipment; wherein the first information comprises time information and/or a connection release reason, the time information is used for indicating that the terminal equipment does not expect to add a time window of the wireless connection, and the connection release reason is used for indicating a reason for releasing the at least one wireless connection.