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 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 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.

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 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.