Children telephone multi-user collaborative talkback method and system

Abstract

The invention discloses a children telephone multi-user collaborative talkback method and a children telephone multi-user collaborative talkback system. Comprising a first terminal used for receiving a talkback event triggered by a user in a talkback group and collecting talkback audio data; the talkback group comprises one or more of a children telephone terminal, a mobile phone terminal and a computer terminal; storing the audio data into an annular buffer area, triggering coding after the frame is full, and reading and coding at one time to obtain coded audio data; sending the coded audio data to a server; the receiving module is also used for receiving coded audio data from a second terminal and sent by the server; decoding the coded audio data of the second terminal to obtain a voice message; and the server is used for receiving the coded audio data sent by the first terminal and forwarding the coded audio data to each second terminal except the first terminal in the talkback group. According to the system provided by the invention, on the basis of the existing voice communication function of the children's telephone and on the premise of not increasing hardware cost, the real-time digital talkback function is expanded, and the problem of multi-person cooperative communication is solved.

Description

Technical Field

[0001] The present application relates to the technical field of mobile phones, and in particular to a method and system for collaborative intercom among multiple users of a children's telephone. Background Art

[0002] Children's phones can provide call, message, and real-time positioning functions, which not only meet the needs of parents to monitor their children, but also have the characteristics of simple operation, long standby time, and anti-addiction, and are becoming more and more popular in the market. Voice calls are the most important function of children's phones. Existing voice technical solutions include: GSM (Global System for Mobile Communications) phones, VoLTE (Voice over Long-Term Evolution) phones, VoIP (Voice over Internet Protocol) Internet phones, and WeChat voice messages. The phone function is real-time communication, while the message function provides delayed communication.

[0003] The existing voice implementation solutions are basically point-to-point communication scenarios, that is, one-to-one voice scenarios between children and parents, but cannot meet the scenarios of multi-person collaborative communication. To expand the function of real-time intercom, if we refer to the intercom implementation solution, the hardware cost will increase, and the whole system solution will be very complicated. Summary of the invention

[0004] The embodiments of the present application provide a method and system for multi-user collaborative intercom on a children's telephone, so as to at least solve the technical problem that multi-user collaborative communication is difficult in the related art.

[0005] According to one aspect of an embodiment of the present application, a method for collaborative intercom of a child phone with multiple users is provided, which is applied to a first terminal and includes:

[0006] Receiving an intercom event triggered by a user in an intercom group, and collecting audio data of the intercom; the first terminal is any terminal in the intercom group; the intercom group includes one or more of a child telephone terminal, a mobile phone terminal, and a computer terminal;

[0007] The audio data is stored in a circular buffer, encoding is triggered after a full frame, and the audio data is read and encoded at one time to obtain the encoded audio data;

[0008] The encoded audio data is sent to a server, so that the server forwards the encoded audio data to each second terminal in the intercom group except the first terminal.

[0009] In one embodiment, it further includes:

[0010] Receiving the encoded audio data sent by the server from the second terminal;

[0011] The encoded audio data of the second terminal is decoded to obtain a voice message.

[0012] In one implementation, the audio data is stored in a ring buffer, encoding is triggered after a full frame, and the encoded audio data is read and encoded at one time, including:

[0013] Storing the collected audio data into a preset ring buffer;

[0014] When the data in the ring buffer reaches a preset number of frames, triggering an encoding task;

[0015] When the encoding task starts, the data of the preset number of frames are read at one time for encoding processing to obtain the encoded audio data.

[0016] In one implementation, sending the encoded audio data to a server includes:

[0017] Packaging the encoded audio data according to a preset service transmission protocol to obtain an audio data packet;

[0018] The audio data packet is sent to the server via the User Datagram Protocol UDP.

[0019] In one embodiment, the protocol frame format of the service transmission protocol includes a start code, a header, service data, a checksum and an end code; the header includes an instruction code, a device number, a packet sequence and a data length;

[0020] The instruction types of the service transmission protocol include: one or more of: heartbeat, login, logout, voice, grab microphone, release microphone, and microphone broadcast.

[0021] In one implementation, decoding the encoded audio data of the second terminal includes:

[0022] Decoding the encoded audio data, and storing the audio data of a preset number of frames decoded each time into a ring buffer;

[0023] A frame of audio data is read from the ring buffer for playback.

[0024] In one implementation, when the first terminal is the mobile phone terminal, the method further includes:

[0025] Sending a group creation request to the server, wherein the group creation request includes guardian information, child phone terminal information, and mobile phone terminal information;

[0026] A group establishment success notification sent by the server when the guardian information is successfully verified is received, and it is determined that an intercom group is established with at least one child telephone terminal and at least one mobile phone terminal.

[0027] According to one aspect of an embodiment of the present application, a method for collaborative intercom of a child phone with multiple users is provided, which is applied to a server and includes:

[0028] Receiving encoded audio data sent by a first terminal, where the first terminal is any terminal in an intercom group, and the intercom group includes one or more of a child telephone terminal, a mobile phone terminal, and a computer terminal;

[0029] The encoded audio data is forwarded to each second terminal in the intercom group except the first terminal based on the transmission control protocol TCP.

[0030] In one embodiment, it further includes:

[0031] Receive a group creation request sent by a mobile terminal;

[0032] Verifying the guardian information of the child's telephone terminal based on the group establishment request;

[0033] After the guardian information is verified, a group establishment success notification is sent to the mobile phone terminal, and the intercom group is established on the guardian's mobile phone terminal.

[0034] According to another aspect of the embodiment of the present application, a children's telephone multi-user collaborative intercom system is provided, comprising:

[0035] The first terminal is used to receive an intercom event triggered by a user in an intercom group and collect audio data of the intercom; the first terminal is any terminal in the intercom group; the intercom group includes one or more of a children's phone terminal, a mobile phone terminal, and a computer terminal; the audio data is stored in a circular buffer, encoding is triggered after a full frame, and the encoded audio data is read and encoded at one time; the encoded audio data is sent to a server, so that the server forwards the encoded audio data to each second terminal in the intercom group except the first terminal; it is also used to receive the encoded audio data sent by the server from the second terminal; the encoded audio data of the second terminal is decoded to obtain a voice message;

[0036] The server is used to receive the encoded audio data sent by the first terminal, and forward the encoded audio data to each second terminal in the intercom group except the first terminal based on the transmission control protocol TCP.

[0037] The technical solution provided by the embodiments of the present application may have the following beneficial effects:

[0038] The multi-user collaborative intercom method of the embodiment of the present application constructs an intercom group, and after receiving an intercom event triggered by a user in the intercom group, collects the audio data of the intercom; encodes the audio data, and forwards the encoded audio data to other terminals in the intercom group based on the server. So that other terminals in the intercom group decode the encoded audio data to obtain voice messages. The intercom group of the present application may include multiple children's phone terminals and multiple mobile phone terminals. Realize multi-person real-time voice communication between multiple children's phone terminals and mobile phone terminals. Moreover, based on the existing voice call function of the children's phone, the present application solution expands the function of real-time digital intercom without increasing the hardware cost, and solves the problem of multi-person collaborative communication. BRIEF DESCRIPTION OF THE DRAWINGS

[0039] The drawings described herein are used to provide a further understanding of the present application and constitute a part of the present application. The illustrative embodiments of the present application and their descriptions are used to explain the present application and do not constitute an improper limitation on the present application. In the drawings:

[0040] Figure 1 is a schematic diagram of a method for collaborative intercom among multiple users on a children's telephone according to an embodiment of the present application;

[0041] Figure 2 is a schematic diagram of a voice data flow according to an embodiment of the present application;

[0042] Figure 3 is a schematic diagram of speech coding and decoding according to an embodiment of the present application;

[0043] Figure 4 is a schematic diagram of a voice cache mode according to an embodiment of the present application;

[0044] Figure 5 is a schematic diagram of a coding and decoding buffer process according to an embodiment of the present application;

[0045] Figure 6 This is a schematic diagram of a speech coding sending process according to an embodiment of the present application;

[0046] Figure 7 is a schematic diagram of a speech receiving and decoding process according to an embodiment of the present application;

[0047] Figure 8 This is a schematic diagram of a multi-user collaborative intercom system for a children's telephone according to an embodiment of the present application. DETAILED DESCRIPTION

[0048] In order to enable those skilled in the art to better understand the solution of the present application, the technical solution in the embodiments of the present application will be clearly and completely described below in conjunction with the drawings in the embodiments of the present application. Obviously, the described embodiments are only part of the embodiments of the present application, not all of the embodiments. Based on the embodiments in the present application, all other embodiments obtained by ordinary technicians in this field without creative work should fall within the scope of protection of the present application.

[0049] It should be noted that the terms "first", "second", etc. in the specification and claims of the present application and the above-mentioned drawings are used to distinguish similar objects, and are not necessarily used to describe a specific order or sequence. It should be understood that the data used in this way can be interchangeable where appropriate, so that the embodiments of the present application described herein can be implemented in an order other than those illustrated or described herein. In addition, the terms "including" and "having" and any of their variations are intended to cover non-exclusive inclusions, for example, a process, method, system, product or device comprising a series of steps or units is not necessarily limited to those steps or units clearly listed, but may include other steps or units that are not clearly listed or inherent to these processes, methods, products or devices.

[0050] The existing voice implementation solutions basically solve point-to-point communication scenarios, that is, one-to-one voice scenarios between children and parents, but cannot meet the scenarios of collaborative communication among multiple people. To expand the function of real-time intercom, if the intercom implementation solution is used, the hardware cost will increase, and the whole system solution will be very complicated. In view of this, the present invention hopes to expand digital voice intercom with pure software on the basis of the existing functions of children's phones without increasing hardware costs, so as to meet the needs of group real-time intercom.

[0051] The following is combined with Figure 1-7 The method for cooperative intercom among multiple users of a children's phone in an embodiment of the present application is described in detail. It should be noted that the following application scenarios are only shown to facilitate understanding of the spirit and principle of the present application, and the implementation of the present application is not limited in this respect. On the contrary, the implementation of the present application can be applied to any applicable scenario. Figure 1 As shown, a method for cooperative intercom among multiple users of a children's telephone is applied to a first terminal. The method mainly comprises the following steps:

[0052] S101 receives an intercom event triggered by a user in an intercom group and collects audio data of the intercom.

[0053] In one embodiment, an intercom group can be pre-built. The intercom group includes one or more of a child's phone terminal, a mobile phone terminal, and a computer terminal. The first terminal is any terminal in the intercom group. The real-time intercom method of the present application can build a real-time intercom group with a child's phone and a parent's mobile phone or computer or other communication terminal.

[0054] In one embodiment, when the first terminal is a mobile phone terminal, the method further includes: sending a group establishment request to a server, the group establishment request including guardian information, child phone terminal information, mobile phone terminal information, etc., and the server verifies the group establishment request. When the guardian information is successfully verified by the server, a group establishment success notification sent by the server is received, and it is determined that an intercom group with at least one child phone terminal and at least one mobile phone terminal is established.

[0055] Through the implementation method of the present application, the guardian of the child's phone can initiate a group creation request to establish an intercom group.

[0056] Furthermore, an intercom event triggered by a user in the intercom group is received, and audio data of the intercom is collected.

[0057] Specifically, the present application expands the digital intercom capability on the children's phone and mobile phone APP, and the phone sends and receives intercom voice data in real time in POC (Push to talk Over Cellular) mode.

[0058] In one implementation, the user triggers a PTT (Push to Talk) event by long pressing a button. After the POC application receives the PTT event, the POC application starts PCM (Pulse-Code Modulation) audio data recording to achieve audio data collection.

[0059] S102 stores the audio data into a circular buffer, triggers encoding after the frame is full, and reads and encodes the encoded audio data at one time to obtain the encoded audio data.

[0060] Before encoding the audio data, the method also includes integrating a voice codec library into the children's telephone terminal and the mobile phone terminal in the intercom group, and implementing codec processing of the voice data based on the same voice codec library.

[0061] In an optional implementation, the cross-platform integrated voice codec library needs to be transplanted across platforms on children's phones (RTOS systems), Android phones or IOS phones to achieve high-quality voice compression. It is necessary to obtain platform audio read and write permissions and design an intermediate layer for the application part to access the library's codec interface.

[0062] This solution uses the Codec2 open source library, with a compression bit rate range of 700-3200 bit / s and a sound quality between LPC-10 and MELP / AMBE. The algorithm is based on speech sound characteristics and implements linear predictive coding (LPC), line-spectral pair conversion (LSP), multi-band excitation (MBE) and quantization processing.

[0063] The encoding and decoding process framework is as follows Figure 3 As shown, the Codec2 open source library is used for voice encoding and decoding. Other encoding and decoding libraries can also be used, and the embodiment of the present application does not make specific limitations.

[0064] The encoder uses data to perform pitch estimation, FFT transformation, and LPC analysis on the speech. After pitch estimation, it performs pitch quantization. After FFT transformation, it performs MBE speech estimation. After LPC analysis, it performs LPC conversion LSP, LSP quantization, LPC correction, and energy quantization.

[0065] The decoder converts the LSPs transmitted by the encoder to LPC, performs FFT transformation on the transmitted energy and LCP, performs magnitude recovery on the transmitted LPC correction, performs phase synthesis and inverse FFT transformation on the speech signal, performs post-filtering on the data after magnitude recovery, and performs inverse FFT transformation after post-filtering. After inverse FFT transformation, overlap synthesis is performed to synthesize speech data.

[0066] The embodiment of the present application adopts a ring buffer. In one implementation, the collected audio data is stored in a preset ring buffer. When the data in the ring buffer reaches a preset number of frames, the encoding task is triggered. When the encoding task starts, the preset number of frames of data is read at one time for encoding processing to obtain encoded audio data. The preset number of frames can be 10 frames, 15 frames, or 18 frames. 、 20 frames, etc., are not specifically limited in the embodiments of the present application.

[0067] Preferably, the embodiment of the present application caches 10 frames of audio data at a time for encoding and decoding.

[0068] The embodiment of the present application connects to the platform PCM (Pulse-Code Modulation) data interface to realize the voice collection and playback functions.

[0069] Specifically, the platform PCM audio sampling data is reported at a frequency of 20ms / frame. If frame-by-frame encoding processing cannot guarantee the real-time encoding efficiency of the data, there are two reasons for this: one is the limitation of the CPU main frequency resources of the children's phone, and the other is the waste of resources caused by frequent task switching. This leads to the problem of delayed frame loss, which seriously affects the integrity of voice playback.

[0070] To solve this problem, the present application designs a data buffer to store each frame of sampling data in a circular buffer. This solution uses 10 frames of data as the minimum buffer length, that is, caching 200ms of sampling data. When 10 frames are full, the encoding task is triggered. 10 frames of data are encoded and sent at a time. If the cached data length still meets 10 frames, encoding continues. If not, the encoding task waits for the sampling task to complete the 10-frame cache.

[0071] Similarly, in the decoding and playback tasks, the circular buffer design is still used. For each packet of compressed data received, 10 frames of PCM data can be decoded and written into the buffer. The playback task obtains one frame of data at a frequency of 20ms and writes it into the playback interface.

[0072] Ring buffer such as Figure 4 As shown in the figure, at the encoding end: Pulse Code Modulation (PCM) is written into the ring buffer at a frequency of 20ms / frame, and 10 frames of data are read at a time for compression and transmission during encoding. At the decoding end: 10 frames of data are decoded at a time and written into the ring buffer during decoding, and one frame of data is read at 20ms during playback of Pulse Code Modulation to complete the playback of voice.

[0073] The read and write tasks of the ring buffer are synchronized by frame length. It has been proven that this design ensures the efficiency of real-time intercom encoding and decoding.

[0074] like Figure 5 As shown, the encoding and decoding buffer process includes:

[0075] During encoding, the voice data transmitted by the PCM interface is received and cached, and it is determined whether 10 frames have been cached. If 10 frames have been cached, the encoding task is notified to perform encoding.

[0076] During decoding, the encoded data is received, the voice is decoded frame by frame, and it is determined whether 10 frames of decoded data have been cached. If 10 frames have been cached, the audio playback system will trigger a callback function every 20ms to process and play the PCM audio data.

[0077] The encoding and decoding method of the present application can improve the real-time performance of voice processing and improve user experience.

[0078] S103 sends the encoded audio data to the server, so that the server forwards the encoded audio data to each second terminal in the intercom group except the first terminal.

[0079] In one embodiment, sending the encoded audio data to the server includes: packaging the encoded audio data according to a preset service transmission protocol to obtain an audio data packet, and sending the audio data packet to the server via a user datagram protocol UDP.

[0080] Among them, the protocol frame format of the service transmission protocol includes a start code, a header, service data, a checksum and an end code; the header includes an instruction code, a device number, a packet sequence and a data length.

[0081] As shown in the following table, the protocol frame format is defined:

[0082]

[0083]

[0084] The command types of the service transmission protocol include: one or more of: heartbeat, login, logout, voice, grab microphone, release microphone, and microphone broadcast.

[0085] The instruction types and corresponding function descriptions are shown in the following table.

[0086] Instruction Type Functional Description Heartbeat Maintain a long connection on the receiving link to ensure that the device is online Log in Enter the group quit Exit Group voice Transmit intercom voice data Grab the microphone Get the right to speak Put wheat Release the right to speak Microphone Broadcast Prompt tone notification to inform group members of microphone occupation status

[0087] The method and protocol of the embodiment of the present application include the following key instruction types, each of which has its specific function:

[0088] Heartbeat: sent periodically to maintain a long connection between the device and the server to ensure the device is online. Login: used to join an intercom group, allowing users to enter a specific group for communication. Exit: used to exit the current intercom group and end communication in the group. Voice: used to transmit intercom voice data and is the core function for achieving real-time voice communication. Grab the microphone: used to obtain intercom speech permissions, allowing users to gain the right to speak in multi-person intercoms. Release the microphone: used to release the intercom speech permissions. When the user finishes speaking, the microphone is released to give other users the opportunity to speak. Microphone broadcast: used to send a prompt tone notification to inform group members of the current microphone occupation status, such as who is speaking.

[0089] The transmission protocol of this application is designed to meet the needs of real-time intercom services. By defining clear command types, the management of intercom groups (such as formation and exit), the transmission of voice data, and the control of intercom speech rights become simple and efficient. Such a protocol design helps to reduce network delays, improve the real-time nature of communications, and is also easy to implement and maintain.

[0090] On the other hand, the audio data packet is sent to the server via the User Datagram Protocol (UDP). The server forwards the encoded audio data to each second terminal in the intercom group except the first terminal based on the Transmission Control Protocol (TCP).

[0091] The transmission design of voice data is crucial to meet the real-time requirements of voice intercom. This solution designs a transmission scheme that sends data in UDP socket mode and receives data in TCP socket mode to reduce network transmission delay and meet data real-time requirements. A method of mixing UDP and TCP sockets is adopted.

[0092] The following is a detailed analysis of the program:

[0093] UDP transmission: Advantages: UDP (User Datagram Protocol) is known for its low latency and high efficiency, and is suitable for real-time applications. It does not require a connection to be established, so the latency of sending data is low. Disadvantages: UDP does not guarantee the order and integrity of data packets, and data packets may be lost. However, in voice transmission scenarios, occasional data packet loss has little impact on user experience, because the human ear is not sensitive to short-term packet loss, and it can be compensated by some algorithms (such as forward error correction).

[0094] Design considerations: In order to reduce the delay of UDP sending, a packet assembly time of 200ms is designed. This ensures the timing of data packets without sacrificing too much real-time performance.

[0095] TCP reception: Advantages: TCP (Transmission Control Protocol) provides reliable data transmission services, ensuring the integrity and order of data packets. This is essential for the receiver to correctly assemble and play voice data. Disadvantages: The TCP connection establishment and maintenance process is relatively complex, which may introduce additional delays. However, on the receiving end, this delay has less impact on the user experience because the main goal of the receiver is to ensure the integrity and correct playback of data.

[0096] Mixed use of UDP and TCP: This design combines the low latency of UDP and the reliability of TCP. UDP is used to send voice data to reduce transmission delay; TCP is used to receive to ensure data integrity and sequence. By introducing a 200ms packet delay at the UDP sender, the timing of data packets can be guaranteed to a certain extent, reducing voice playback anomalies caused by network fluctuations.

[0097] Performance under poor network conditions: The solution is designed with poor network conditions in mind. By introducing appropriate delays on the UDP sender, acceptable voice quality can be maintained even when network conditions are poor.

[0098] In general, this hybrid UDP and TCP transmission scheme provides an effective network transmission solution for real-time voice intercom applications by balancing transmission efficiency and data integrity.

[0099] In one implementation, the method further includes: receiving encoded audio data from the second terminal sent by the service end; and decoding the encoded audio data of the second terminal to obtain a voice message.

[0100] Any terminal in the intercom group can send and receive voice coded messages. For the first terminal, the coded audio data from the second terminal sent by the server can be received; the coded audio data of the second terminal is decoded to obtain a voice message. The second terminal is any other terminal in the intercom group except the first terminal.

[0101] Decode the encoded audio data, and store the preset number of frames of audio data in the ring buffer each time; read a frame of audio data from the ring buffer for playback. This application still uses the design of a ring buffer when decoding the encoded speech. Each packet of the received compressed data can be decoded into 10 frames of PCM data and written into the buffer. The playback task obtains a frame of data at a frequency of 20ms and writes it to the playback interface.

[0102] According to one aspect of an embodiment of the present application, a method for collaborative intercom of a child phone with multiple users is provided, which is applied to a server and includes:

[0103] Receive the encoded audio data sent by the first terminal, the first terminal is any terminal in the intercom group, and the intercom group includes one or more of children's telephone terminals, mobile phone terminals, and computer terminals; forward the encoded audio data to each second terminal in the intercom group except the first terminal based on the transmission control protocol TCP.

[0104] In one embodiment, it also includes: receiving a group establishment request sent by a mobile phone terminal; verifying the guardian information of the child's phone terminal based on the group establishment request; after the guardian information is verified, sending a group establishment success notification to the mobile phone terminal, and establishing an intercom group on the guardian's mobile phone terminal.

[0105] Receiving and forwarding encoded data is realized based on the server.

[0106] like Figure 2 As shown, the voice data flow includes obtaining voice sampling data, caching data, encoding and compressing the cached data, protocol packaging of the compressed data, and then data transmission based on the server. At the voice data receiving end, protocol unpacking, decoding and restoration, voice caching, and finally voice playback are performed.

[0107] Its overall business process includes:

[0108] In the real-time intercom system, when the user triggers the intercom button event, the POC application immediately starts the PCM sampling task and collects audio data at a frequency of 20ms per frame. This high-frequency collection ensures the real-time nature of the audio data and meets the strict requirements for low latency in intercom scenarios.

[0109] The collected PCM data is first stored in the buffer. Once a sufficient number of frames (e.g. 10 frames) are accumulated in the buffer, the POC application notifies the encoding task to start working. The encoding task is responsible for reading these PCM data and performing audio compression processing to reduce the amount of data for network transmission. The compressed audio data is packaged according to the network protocol. This step is to ensure the efficiency and correctness of data transmission in the network.

[0110] After the packetization is completed, the audio data is sent to the server via the UDP (User Datagram Protocol). UDP is selected for its low latency and high transmission efficiency. Although it does not guarantee reliable data transmission, in intercom scenarios, occasional data packet loss has little impact on user experience and can be compensated by other technical means.

[0111] After receiving the UDP data packet, the server is responsible for forwarding the group data. To ensure the integrity and order of the data, the server uses the TCP (Transmission Control Protocol) protocol to send the data to other terminal devices in the group, such as children's phones and mobile APPs. The reliability of the TCP protocol ensures the stability of the data during transmission, and can ensure the correct assembly and playback of audio data even under poor network conditions.

[0112] The entire process design fully considers the characteristics of real-time intercom applications. By making reasonable use of the high efficiency of UDP and the reliability of TCP, low-latency and high-stability audio transmission is achieved, meeting the user's demand for real-time intercom experience. At the same time, through the careful design of the client and server, it ensures that even in complex and changeable network environments, smooth and clear intercom services can be provided.

[0113] In order to facilitate understanding of the business process of the embodiment of the present application, Figure 6 and 7 Further description.

[0114] like Figure 6 As shown, the speech coding sending process includes:

[0115] The user triggers the PTT event by long pressing the button. After the POC application receives the PTT event, it starts recording PCM audio data. The PCM sampling task collects audio data at a frequency of 20ms per frame and writes the data into the ring buffer. The encoding task monitors the ring buffer, and once new audio data is written, it reads the data from the ring buffer. The encoding task performs audio compression on the read PCM data and then packages it with the transport protocol. The packaged audio data is sent out via the UDP protocol to achieve real-time audio transmission.

[0116] like Figure 7 As shown, the voice receiving and decoding process includes:

[0117] The decoding task first performs transport protocol unpacking and audio decoding on the received data, and then writes the decoded PCM audio data into the ring buffer. The PCM playback task reads this data from the ring buffer and performs callback playback at intervals of 20ms, thereby achieving continuous playback of audio data. The entire process ensures the real-time and smoothness of audio data from reception, decoding to playback.

[0118] This application scheme introduces a new voice communication method for children's phones through innovative software design, significantly improving the market competitiveness of the product. The core of this scheme is its support for group multi-person collaborative intercom function, which not only enriches the communication mode of children's phones, but also meets the needs of multi-person real-time communication in scenarios such as home and school.

[0119] In terms of technical implementation, this solution cleverly utilizes existing hardware resources without additional hardware investment, and realizes efficient digital voice intercom function through software optimization and protocol design. This design concept not only reduces costs, but also ensures the real-time and clarity of voice communication, so that children's phones can provide a communication experience comparable to traditional intercoms without increasing hardware costs.

[0120] According to another aspect of the embodiment of the present application, a children's phone multi-user collaborative intercom system for implementing the above-mentioned children's phone multi-user collaborative intercom method is also provided. It includes:

[0121] The first terminal is used to receive an intercom event triggered by a user in the intercom group and collect audio data of the intercom; the first terminal is any terminal in the intercom group; the intercom group includes one or more of a child phone terminal, a mobile phone terminal, and a computer terminal; the audio data is stored in a circular buffer, encoding is triggered after a full frame, and the encoded audio data is read and encoded at one time; the encoded audio data is sent to the server, so that the server forwards the encoded audio data to each second terminal in the intercom group except the first terminal; it is also used to receive the encoded audio data sent by the server from the second terminal; the encoded audio data of the second terminal is decoded to obtain a voice message;

[0122] The server is used to receive the encoded audio data sent by the first terminal, and forward the encoded audio data to each second terminal in the intercom group except the first terminal based on the transmission control protocol TCP.

[0123] like Figure 8 As shown in the figure, the system includes: server, POC application and multiple client devices, such as children's phones and mobile phone apps. The server is the core of the system and is responsible for managing and forwarding voice data within the group. The POC application serves as a bridge connecting the server and the client to realize the collection, encoding, compression and transmission of voice data.

[0124] The user joins the intercom group through a child phone or mobile phone APP. The POC application installed on these devices is responsible for collecting the user's voice input. When the user triggers the intercom button on the client device, the POC application collects PCM audio data at a frequency of 20ms per frame. The collected audio data is first cached, and when it reaches a certain amount (such as 10 frames), the POC application notifies the encoding task to start working. The encoding task compresses this data to reduce the amount of data for network transmission. The compressed audio data is packaged according to the network protocol and sent to the server via the UDP protocol. UDP is selected to reduce network transmission delays and meet the needs of real-time intercom. After receiving the UDP data packet, the server is responsible for forwarding the group data. To ensure the integrity and sequence of the data, the server uses the TCP protocol to send the data to other terminal devices in the group. Other client devices receive the data forwarded by the server through the TCP protocol, and then decode and play it to achieve real-time intercom.

[0125] It should be noted that the multi-user collaborative intercom system for a children's phone provided in the above embodiment only uses the division of the above-mentioned functional modules as an example when executing the multi-user collaborative intercom method for a children's phone. In actual applications, the above-mentioned functions can be assigned to different functional modules as needed, that is, the internal structure of the device can be divided into different functional modules to complete all or part of the functions described above. In addition, the multi-user collaborative intercom system for a children's phone provided in the above embodiment and the multi-user collaborative intercom method embodiment for a children's phone belong to the same concept, and the implementation process thereof is detailed in the method embodiment, which will not be repeated here.

[0126] The technical features of the above embodiments may be combined arbitrarily. To make the description concise, not all possible combinations of the technical features in the above embodiments are described. However, as long as there is no contradiction in the combination of these technical features, they should be considered to be within the scope of this specification.

[0127] The above embodiments only express several implementation methods of the present invention, and the descriptions thereof are relatively specific and detailed, but they cannot be understood as limiting the scope of the present invention. It should be pointed out that, for a person of ordinary skill in the art, several modifications and improvements can be made without departing from the concept of the present invention, and these all belong to the protection scope of the present invention. Therefore, the protection scope of the present invention patent shall be subject to the attached claims.

Claims

1. A method for cooperative intercom of multiple users on a children's telephone, characterized in that: Applied to the first terminal, comprising: Receiving an intercom event triggered by a user in an intercom group, and collecting audio data of the intercom; the first terminal is any terminal in the intercom group; the intercom group includes one or more of a child telephone terminal, a mobile phone terminal, and a computer terminal; The audio data is stored in a circular buffer, encoding is triggered after a full frame, and the audio data is read and encoded at one time to obtain the encoded audio data; The encoded audio data is sent to a server, so that the server forwards the encoded audio data to each second terminal in the intercom group except the first terminal.

2. The method according to claim 1, characterized in that Also includes: Receiving the encoded audio data sent by the server from the second terminal; The encoded audio data of the second terminal is decoded to obtain a voice message.

3. The method according to claim 1, characterized in that The audio data is stored in a ring buffer, and encoding is triggered after a full frame, and the encoded audio data is read and encoded at one time, including: Storing the collected audio data into a preset ring buffer; When the data in the ring buffer reaches a preset number of frames, triggering an encoding task; When the encoding task starts, the data of the preset number of frames are read at one time for encoding processing to obtain the encoded audio data.

4. The method according to claim 1, characterized in that Sending the encoded audio data to the server includes: Packaging the encoded audio data according to a preset service transmission protocol to obtain an audio data packet; The audio data packet is sent to the server via the User Datagram Protocol UDP.

5. The method according to claim 4, characterized in that The protocol frame format of the service transmission protocol includes a start code, a header, service data, a checksum and an end code; the header includes an instruction code, a device number, a packet sequence and a data length; The instruction types of the service transmission protocol include: one or more of: heartbeat, login, logout, voice, grab microphone, release microphone, and microphone broadcast.

6. The method according to claim 2, characterized in that Decoding the encoded audio data of the second terminal includes: Decoding the encoded audio data, and storing the audio data of a preset number of frames decoded each time into a ring buffer; A frame of audio data is read from the ring buffer for playback.

7. The method according to claim 1, characterized in that When the first terminal is the mobile phone terminal, the method further includes: Sending a group creation request to the server, wherein the group creation request includes guardian information, child phone terminal information, and mobile phone terminal information; A group establishment success notification sent by the server when the guardian information is successfully verified is received, and it is determined that an intercom group with at least one child telephone terminal is established.

8. A method for cooperative intercom of multiple users on a children's telephone, characterized in that: Applied to the server, including: Receiving encoded audio data sent by a first terminal, where the first terminal is any terminal in an intercom group, and the intercom group includes one or more of a child telephone terminal, a mobile phone terminal, and a computer terminal; The encoded audio data is forwarded to each second terminal in the intercom group except the first terminal based on the transmission control protocol TCP.

9. The method according to claim 8, characterized in that Also includes: Receive a group creation request sent by a mobile terminal; Verifying the guardian information of the child's telephone terminal based on the group establishment request; After the guardian information is verified, a group establishment success notification is sent to the mobile phone terminal, and the intercom group is established on the guardian's mobile phone terminal.

10. A children's telephone multi-user collaborative intercom system, characterized in that: include: The first terminal is used to receive an intercom event triggered by a user in the intercom group and collect audio data of the intercom; The first terminal is any terminal in the intercom group; the intercom group includes one or more of a child phone terminal, a mobile phone terminal, and a computer terminal; the audio data is stored in a ring buffer, encoding is triggered after a full frame, and the encoded audio data is read and encoded at one time; the encoded audio data is sent to the server, so that the server forwards the encoded audio data to each second terminal in the intercom group except the first terminal; and it is also used to receive the encoded audio data sent by the server from the second terminal; decoding the encoded audio data of the second terminal to obtain a voice message; The server is used to receive the encoded audio data sent by the first terminal, and forward the encoded audio data to each second terminal in the intercom group except the first terminal based on the transmission control protocol TCP.

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