30results about How to "Smooth call" patented technology

The invention discloses a method for processing audio stream playback in the network terminal buffer zone for solving the problem of voice pausing and jamming in the network communication. The aim of the invention is achieved by performing real time audio (e.g. voice) communication on the packet-switching network (e.g. IP network), arranging a shake buffer zone on the receiving end, after the receiving end receives the audio package, it first performs decoding based on the normal sequence, then places it into the shake buffer zone, when the shake buffer zone is to be filled, lower the sampling rate to the audio data to realize the fast playback of the audio data stream, when the shake buffer zone is to be empty, raise the sampling rate to the audio data to realize the low speed playback of the audio data stream, when the audio data in the shake buffer is within the normal range, playback the audio stream with the original sampling rate.

The invention is applicable in the field of communication and provides a communication noise-reducing circuit and a noise-reducing method. The communication noise-reducing circuit comprises a main microphone, a secondary microphone, a differential amplifier, a mixed amplifier, a comparator and an analog switch, wherein the mixed amplifier is used for receiving audio signals acquired by the main microphone and the secondary microphone and performing mixed amplification processing; the comparator acquires an audio signal which is subjected to differential amplification processing by the differential amplifier, compares the audio signal subjected to differential amplification processing with a preset threshold signal to obtain a comparison result and outputting the comparison result to the analog switch; and the analog switch receives the comparison result output by the comparator and selects the audio signal subjected to differential amplification processing or the audio signal subjected to mixed amplification processing according to the comparison result to output. The communication noise-reducing circuit can automatically identify a communication environment to perform noise-reducing processing, so the communication noise-reducing circuit is very convenient for a user to use.

The embodiment of the invention discloses a method for switching cells. The method comprises the following steps: dividing micro cells under the coverage of signals of macro cell base stations into atleast two clusters, distributing same phase bias groups of pseudo random codes to each cluster, determining target micro cells in each cluster based on the phase bias of the pseudo random codes carried by the terminal switching requests, distributing public traffic channels to the micro cells to switch cells, and switching the terminal from the public traffic channels to common traffic channels.The embodiment of the invention also provides a cell switching device and a communication system thereof, solving the problems of failures, call drops and the like of terminal talk switching as a result of not merging of useful lists due to overlong neighbor lists.

The invention provides a method and system for switching the microphone function of a mobile terminal. Whether new incoming call is received by the mobile terminal is detected. When the mobile terminal receives a new incoming call, the location angle information of the mobile terminal is obtained, and whether the mobile terminal is inverted is determined according to the location angle information obtained. If the mobile terminal is inverted, the function of a first microphone on the upper end portion of the mobile terminal is switched from noise signal reception to voice signal reception, and the function of a second microphone on the lower end portion of the mobile terminal is switched from voice signal reception to noise signal reception. When the mobile terminal held in a user's hand is inverted, the mobile terminal still can be used for smooth communication. The method and system bring convenience to users.

The invention discloses a method and apparatus for processing simultaneous mutual call. The method comprises: a) calling a second telephone number using a first telephone number; b) detecting whether an other communication terminal calls the first telephone number at the same time by using the second telephone number; c) selecting the first telephone number or the second telephone number in a predetermined manner when the second telephone number calls the first telephone number at the same time; d)reminding a user to call again the second telephone number or controlling the communication terminal to call again the second telephone number when the first telephone number is selected; and e) when the second telephone number is selected, reminding the user to wait for the calling of the second telephone number and / or prohibiting calling again the second telephone number during a first predetermined time period.

The present invention discloses a mobile communication terminal with noise-identification communication-variation function and method of varying the same, characterized by comprising parts of: a display part capable of converting received voice signal in to character and outputting the same; a control module automatically measuring and determining surrounding noise of receiver in communication, if noise having noise value higher than set reference, converting received voice signal in to character and outputting the same onto said display part. The invention is different with method of amplifying communication voice of existing art, and has effects that, even surrounding noise exists in communication process, voice of opposite side can be outputted as character message on LCD display screen, such that voice of opposite side can be comprehended indeed and recognized in vision, which achieves continuous and stable communication, and prevents generating damnification in hearing.

The invention discloses a sound input / output system and a volume output adjusting method thereof, which are used for automatically adjusting the volume of sound listened by a speaker according to the speaking sound of the speaker. The sound input / output system comprises a sound processor and a volume adjuster, wherein the sound processor is used for resolving the input sound received by a receiver, and the volume adjuster is used for adjusting the output sound of a loudspeaker according to a sound change value acquired by the sound processor. When the speaker changes the speaking volume alongwith the environment or the listened input sound, the sound processor can resolve the sound change value of the input sound, and then adjust the volume of the output sound according to the sound change value.

The invention provides a voice transmission method. The voice transmission method comprises the following steps: receiving a voice call transmission command sent by a first terminal, acquiring voice information to be transmitted according to the voice call transmission command, and receiving the voice information to be transmitted by a second terminal; acquiring the transmission rate of the voiceinformation to be transmitted; judging whether the transmission rate is lower than a preset transmission rate or not; if the transmission rate is lower than the preset transmission rate, performing voice recognition on the voice information to be transmitted to obtain a voice recognition result, wherein the voice recognition result includes text information corresponding to the voice information to be transmitted; performing voice coding on the text information included in the voice recognition result to obtain target voice information; and transmitting the target voice information to the second terminal. The invention further discloses a voice transmission device, a computer device and a computer readable storage medium. Through adoption of the voice transmission method and device, the voice call quality can be enhanced.

The invention discloses a method for implementing full-duplex group calling in a digital trunked communication system. The method comprises the steps of identifying and establishing the full-duplex group calling in the digital trunked communication system; processing speaking of a calling party and group members in a calling process. When a mobile terminal subscriber in the system dials a full-duplex group calling number of a trunked system, a control channel of the trunked system allocates two idle service channels to the calling; one of the service channels is allocated to a calling subscriber and works in a full-duplex mode, so that the calling subscriber can directly speak without a speaking authorization applying process during speaking; the other service channel is allocated to the group members and works in a half-duplex mode, so that the group members need to apply speaking authorization during speaking and can speak with permission of the system; after the speaking is finished, a speaking party releases the speaking authorization. After the whole calling is finished, the two service channels simultaneously release the speaking authorization, and the advantages of a trunked private network scheduling system can be taken to the greatest extent.

The invention provides a call control method and a mobile terminal, and relates to the technical field of communication. The call control method comprises the following steps: in a call process, detecting whether a current mode is an earphone call mode or not; if the current mode is detected to be the earphone call mode, detecting whether a current call state satisfies a preset privacy call mode starting condition or not; and if the current call state is detected to satisfy the preset privacy call mode starting condition, controlling a preset region of the mobile terminal to bend, wherein thepreset region is provided with a microphone. Through the scheme of the invention, the problem of risk of privacy leakage in an existing call way is solved.

The invention discloses a video call process voice acquisition method and device, computer equipment and a medium. The method comprises the steps: creating a special communication channel associated with a terminal ID between a user terminal and a virtual image platform based on a video call request; receiving call media stream data of the user terminal, associatively storing the target agent ID and the terminal ID, and obtaining a response verbal skill text corresponding to the audio media stream data; sending the response verbal skill text and the target seat ID to a virtual image platform;establishing MRCP connection with the media stream processing platform, and sending a target agent ID associated with the response verbal skill text to the media stream processing platform; if a verbal skill broadcast end notification of a target virtual seat corresponding to the target seat ID sent by the media stream processing platform is received, closing the MRCP connection with the media stream processing platform, starting the voice recognition platform to collect audio media stream data according to the target seat ID, and repeatedly executing receiving of the call media stream data generated by the user terminal to eliminate noise and receive the radio accurately.

The invention discloses a communication device and an input interface automatic locking method thereof. The method firstly obtains external sound volume received by the communication device from the communication device in a communication state, then compares the external sound volume with a sound volume presetting value, closes the input function of the input interface of the communication device if the external sound volume reaches the sound volume presetting value. Thus the invention prevents users from turning on other functions because of accidental touching the input interface during the communication process.

The embodiment of the invention provides a method and device for initiating a Qchat call, and a Qchat call mode switching method and device, and relates to the field of communication. The methods and the devices are used for solving the problem that a selected call mode is not adaptive to a current network environment in a Qchat call initiating process or in a Qchat call process. The method for initiating the Qchat call comprises following steps of determining a communication network demanded for initiating the Qchat call after a one-to-one Qchat call request is received by a mobile terminal; obtaining the communication quality parameters of the communication network; initiating a call of a Qchat instant call mode if the communication quality parameters of the communication network satisfy preset conditions, wherein the preset conditions are the lowest standards of allowing to carry out the Qchat instant call in the communication network; and initiating a call of a Qchat intercom call mode if the communication quality parameters of the communication network do not satisfy the preset conditions.

The invention provides a social cellphone. The cellphone can employ more than one communication mode for calling an opposite side at a time or automatically employs more than one communication modes in an ordered manner to call the opposite side till the opposite side is connected. The cellphone can receive the simultaneous calls from the opposite side in more than one mode, and can achieve the answering of the calls at a time. The cellphone can employ more than one communication mode at a time to transmit information to the opposite side, or automatically employs more than one communication mode in an ordered manner to transmit the information to the opposite side till the opposite side receives or browses the information. The cellphone can receive the information transmitted by the opposite side in more than one communication mode at the same time, and can receive the information at a time. A plurality of interactive window sub-interfaces employed in the above communication process can exist at the same time, and can belong to the same or different application programs and also can exist with other interfaces at the same time. The cellphone can achieve the smoother communication between the people, brings convenience to a holder to communicate with multiple persons, and can meet the social demands of the people in a better way.

The voice service servo-server of the present invention can be used to register network telephone and also can be used to record network actual position being distributed to the network telkephone tomake call between the network telephones connected smoothly. In addition, the calling package with telephone number of another network telephone will be transmitted by the network telephone accordingto transmission control protocol when the network telephone sends "registered package" to voice service servo-server and calls another network telephone, in order to let the voice service servo-server record the effective network actual position.

The invention discloses a communication device and an input interface automatic locking method thereof. The method firstly obtains external sound volume received by the communication device from the communication device in a communication state, then compares the external sound volume with a sound volume presetting value, closes the input function of the input interface of the communication device if the external sound volume reaches the sound volume presetting value. Thus the invention prevents users from turning on other functions because of accidental touching the input interface during the communication process.

The present invention discloses a mobile communication terminal with noise-identification communication-variation function and method of varying the same, characterized by comprising parts of: a display part capable of converting received voice signal in to character and outputting the same; a control module automatically measuring and determining surrounding noise of receiver in communication, if noise having noise value higher than set reference, converting received voice signal in to character and outputting the same onto said display part. The invention is different with method of amplifying communication voice of existing art, and has effects that, even surrounding noise exists in communication process, voice of opposite side can be outputted as character message on LCD display screen, such that voice of opposite side can be comprehended indeed and recognized in vision, which achieves continuous and stable communication, and prevents generating damnification in hearing.

The invention discloses a method for implementing full-duplex group calling in a digital trunked communication system. The method comprises the steps of identifying and establishing the full-duplex group calling in the digital trunked communication system; processing speaking of a calling party and group members in a calling process. When a mobile terminal subscriber in the system dials a full-duplex group calling number of a trunked system, a control channel of the trunked system allocates two idle service channels to the calling; one of the service channels is allocated to a calling subscriber and works in a full-duplex mode, so that the calling subscriber can directly speak without a speaking authorization applying process during speaking; the other service channel is allocated to the group members and works in a half-duplex mode, so that the group members need to apply speaking authorization during speaking and can speak with permission of the system; after the speaking is finished, a speaking party releases the speaking authorization. After the whole calling is finished, the two service channels simultaneously release the speaking authorization, and the advantages of a trunked private network scheduling system can be taken to the greatest extent.

The present disclosure relates to a method for invoking a software development kit, applicable to a platform, including: triggering a target application to invoke a software development kit of a third-party application according to a received preset instruction, wherein the target application runs on the platform; The identification information of the target application is transmitted to the software development kit; the metadata required by the third-party application to verify the target application is determined, and the association between the metadata and the identification information is established; the metadata is registered in the system where the platform is located , so that the third-party application can obtain the metadata related to the identification information according to the association relationship for verification. Based on the embodiments of the present disclosure, it can be ensured that the software development kit of the third-party application can obtain the metadata required for verification from the system according to the identification information for verification, thereby ensuring the smooth calling of the software development kit of the third-party application .

The invention provides a mobile terminal and a method for falling back to a cell. The method includes that the mobile terminal receives the paging request sent by the MME, and sends an extended service request to the MME; the MME instructs the eNB to send configuration information of the fallback cell to the mobile terminal; the mobile terminal receives the configuration information of the fallback cell, and measures the configured fallback cell, And send the measurement results to eNB; eNB processes the measurement results to obtain the best fallback cell; the best fallback cell is the fallback cell that allows access and has the best signal; eNB instructs the mobile terminal to fall back to the best fallback cell; mobile The terminal falls back to the best fallback cell according to the instruction. The method can avoid the situation that the mobile terminal is rejected because the fallback cell is in a congested state or other poor communication state, thereby ensuring that the mobile terminal can successfully fall back, establish a call smoothly, and improve user experience.

The invention belongs to the technical field of medical treatment, and particularly relates to a shelter consulting room for receiving respiratory tract fulminating infectious disease patients, and the end part of an air inlet is provided with a cleaning system; a transition compartment is formed in the end part of the shelter consulting room body; an air outlet is formed in the end part of the shelter consulting room body; all medical consumables which are in contact with patients, including oversleeve type gloves and oversleeve type diaphragms for stethoscopes, are disposable, so that cross infection among the patients is avoided; the upper end of the side surface of the cabin is transparent, so that doctors and patients can see each other; communication with a patient is achieved through the intercom device, firstly, the outer side face of a circular ring at the tail end of a new oversleeve type glove / stethoscope oversleeve type diaphragm is attached to the inner side face of a circular ring of a newly-used polluted glove / diaphragm from a circular hole in a cabin, and then the used polluted glove / diaphragm is torn off; in this way, it can be guaranteed that the oversleeve hole is closed when the polluted glove / diaphragm is broken.

The present application discloses a method for processing voice communication by a mobile terminal, which includes the following steps. Step S10: When the voice communication starts, preset the positions of the read pointer and write pointer in the playing buffer of the receiving end, set a reference value and a threshold; preset the read pointer before the write pointer; the threshold is smaller than the reference value. Step S20: Detecting the distance between the read pointer and the write pointer in the playback buffer every first time is called the detection distance. Step S30: Use the offset state to represent the historical state of the playback buffer, shift the offset state by two bits to the left, and determine a new value for the offset state according to the comparison result between the detection distance and the sum or difference between the reference value and the threshold. Step S40: According to the detection situation of the VAD from the voice PCM data stream, the state of the joint offset determines whether to perform sample number interpolation or extraction. This application is used to solve the impact of time delay on voice communication.

The invention discloses a mobile phone with a sound-guiding microphone structure at the bottom. The sound-guiding hole of the microphone originally arranged on the front of the mobile phone is changed to the bottom surface of the mobile phone. The hole is blocked by the face or the hand, and the microphone of the mobile phone cannot receive the voice signal. The sound guide structure at the bottom is suitable for promotion and adoption of all handheld communication terminals.

The present invention discloses a mobile terminal control method, a mobile terminal and a computer readable storage medium. The method comprises: detecting a network standard of a resident network inreal time; if the network standard of the resident network is a low-level network standard, turning off packet switching ps domain networking function; determining a service list of services affectedby service data interruption when the ps domain networking function is turned off; when detecting that the network standard of the resident network is reselected or switched to an advanced network standard, enabling the ps domain networking function to complete each service in the service list. The method can ensure smooth running of mobile terminal communication, and can also realize automatic recovery of the affected service.