Abstract: The disclosure discloses a voice signal enhancing method and device, which divide a voice signal at the present scene into multiple frame signals based on a preset time interval; feed multiple frame signals into a trained neural network based on a preset step size, perform convolution operations on multiple frame signals through skip-connected convolutional layers to obtain multiple enhanced frame signals; superpose each enhanced frame signal according to the time domain of each enhanced frame signal to obtain an enhanced voice signal. Compared with the prior art, the present disclosure automatically enhances voice signals through the neural network without manual interference, so the effects and the application scenes of voice enhancement is not necessary to be limited by the preset method and method designers, thereby reducing the occurrence frequency of signal distortion and extra noises, which in turn improves the effects of the voice signal enhancement.

Abstract: The present disclosure provides a secondary-stream data transmission method for use in a conference. An electronic device is communicatively connected to a peripheral device, and a virtual display apparatus and a virtual UAC device are virtualized on the electronic device. The virtual UAC device is configured to establish an audio data channel between the electronic device and the peripheral device, to obtain audio data. Image data is obtained from an output interface of a graphics card of the electronic device. If the image data needs to be transmitted to the virtual display apparatus, the image data, or the image data and the audio data are transmitted to the peripheral device, and are transmitted to the outside by the peripheral device. The present disclosure can not only implement compatibility of cross-platform operating systems, but also support different display modes such as duplication and extension.

Abstract: The present disclosure discloses a method and system for generating mixed voice data, and belongs to the technical field of voice recognition. In the method for generating mixed voice data according to the present disclosure, a pure voice and noise are collected first, normalization processing is performed on the collected voice data, randomization processing is performed on processed data, then GAIN processing is performed on the data, and finally filter processing is performed to obtain mixed voice data. The system for generating mixed voice data according to the present disclosure includes a collecting unit, a calculating unit, and a storage unit, the collecting unit being electrically connected to the calculating unit, and the calculating unit being connected to the storage unit through a data transmitting unit. The present disclosure provides the method and the system to meet the data requirement of deep learning.

Abstract: Disclosed are a communication device having a base station function of a Digital Enhanced Cordless Telecommunications (DECT) cellular system, and a communication system. The communication device includes an IP phone and a DECT adapter. The IP phone is connected to the DECT adapter, to form a base station; and the DECT adapter is configured to wirelessly communicate with at least one cordless handset via DECT, so as to register information of the at least one cordless handset with the IP phone. The present application enables the IP phone to be externally connected to the DECT adapter to support a DECT mode, so that the IP phone possesses a base station function of a DECT cellular system. It is not required to deploy multiple different systems. Such an IP phone and other base stations can form a Multi-Cell DECT system, expanding the coverage of a DECT signal.

Abstract: A DECT network clustering method includes sending, by a DECT host to a router, a multicast command that declares joining a specified multicast group; sending, by a DECT device to the router, a multicast probe message sent to a target multicast group; receiving, by the router, the multicast command and the multicast probe message, and sending, to the DECT host according to the multicast command, the multicast probe message; receiving, by the DECT host, the multicast probe message, and sending status declaration information to the DECT device; receiving, by the DECT device, the status declaration information, and initiating a TCP connection request to the DECT host in a point-to-point manner; and receiving, by the DECT host, the connection request from the DECT device, and establishing a TCP channel with the DECT device.

Abstract: The present application discloses a media interaction method in a Digital Enhanced Cordless Telecommunications (DECT) network cluster. Communication with a DECT host is established through a DECT network cluster constructed in advance, to obtain first account information allocated by the DECT host, and information of a first call request to call a second account is sent to the DECT host, so as to make a call to the second account in a first communication manner. If no Real-time Transport Protocol (RTP) from the second account is received, information of a second call request to call the second account is sent to the DECT host, so as to make a call to the second account in a second communication manner. Embodiments of the present disclosure can solve the technical problem of a communication failure and reduce a waiting time for communication, thereby achieving optimal Voice over Internet Protocol (VOIP) call experience.

Abstract: The present disclosure discloses a method, an apparatus and a device for erasing handwriting on an electronic whiteboard. The method includes the following steps: obtaining first contact points; enabling an erasing mode when a quantity of the first contact points is greater than a preset value; obtaining second contact points in the erasing mode, and calculating an erasing region according to the first contact points and the second contact points; and deleting handwriting data in the erasing region. Using the method for erasing handwriting on an electronic whiteboard provided by the present disclosure can enter an erasing mode rapidly and erase handwriting effectively, thus improving the practicability of the electronic whiteboard.

Abstract: The present disclosure discloses a method and a system for automatically resuming a call after network handover, and belongs to the field of network calls. In the method of the present disclosure, during a call between a first terminal and a second terminal, when the first terminal is handed over from a first network to a third network, the first terminal first stores call resources, then registers with a server through the third network, constructs a resume request, and sends the resume request to the second terminal via the server, and the second terminal resumes the call with the first terminal according to the resume request. The present disclosure can ensure communication stability after network handover of a communication device, thereby avoiding call interruption caused by the network handover during communication of the communication device.

Abstract: The present disclosure relates to the field of communications technologies, and specifically, to a video conference terminal and a video conference system. The video conference terminal is provided with a content receiving module (101), an audio and video processing module (102), and a display module (103). To be specific, the video conference terminal (100) has a video conference function, so that the video conference terminal can serve as both an audio conference terminal and a video conference terminal, thereby enriching functions of the video conference terminal. Compared with a conventional conference terminal, the video conference terminal provided in embodiments of the present disclosure has easy deployment, low costs and high utilization as it can serve as a video conference terminal.

Abstract: The present disclosure discloses a method for preventing a SIP device from being attacked, a calling device, and a called device, belonging to the field of network security. The present disclosure provides a method including: generating, by a calling device and a called device, the same public password, and transmitting, by the calling device, a connection request to the called device; performing, by the called device, header field verification on the connection request to verify whether a specified header field is carried in the connection request; performing, by the called device, device verification on the connection request; and performing, by the called device, identity verification on the connection request, and establishing, by the called device, a connection to the calling device. In this case, spoofing data is filtered out and the SIP device is not easily attacked, so that a user is free of disturbance.

Abstract: The present disclosure discloses a multi-device recording synchronization method and system, and a conference system, and belongs to the field of recording marking. For the problem in the prior art that a device that supports a recording marking function only supports generation and use on a single device, the present disclosure provides a system, including one master and a plurality of slaves. The master is connected to the plurality of slaves. The master operates according to the instruction sent by the slave and broadcasts state information globally. Through a network connection between the master and the slave, the master stores, on the master, recording and marking data sent by the slave, and synchronizes the data between devices, and a multi-device recording marking and synchronization system is applied to a conference system. The present disclosure is very convenient for use and has simple operation logic and low system costs.

Abstract: The present disclosure discloses a method for constructing a voice detection model and a voice endpoint detection system, and belongs to the technical field of voice recognition. In the method for constructing a voice detection model according to the present disclosure, audio data is first collected and a mixed voice is synthesized, feature extraction is performed on the mixed voice to obtain a 62-dimensional feature, and then the 62-dimensional feature is input to a recurrent neural network (RNN) model for training to obtain a voice detection model. The voice endpoint detection system according to the present disclosure includes a collecting unit, a calculating unit, a transmitting unit, and a terminal, the collecting unit being electrically connected to the calculating unit, and the calculating unit and the terminal being respectively connected to the transmitting unit. The voice detection model can be applied to a real-time conference communication device.

Abstract: The embodiments of the present disclosure disclose a firmware boot implementation method based on Flash chip simulation, the method comprising: when there are at least two MCUs, one of the MCUs is selected as a master MCU and each remaining MCU is as slave MCU which is connected with the master MCU respectively, and the master MCU is connected with a Flash chip; when the master MCU is started, the master MCU is started by reading firmware data in the Flash chip; when the slave MCU is started, the master MCU controls the slave MCU to be powered on, and the slave MCU sends a request for reading the firmware data to the master MCU; the master MCU transmits the request to the Flash chip, and transmits the firmware data returned by the Flash chip to the slave MCU, so as to start the slave MCU.