Many-to-many voice communication system and method, electronic equipment and storage medium

Abstract

The invention relates to a many-to-many voice communication system and method, an electronic device and a storage medium, and the system comprises a plurality of voice terminal devices, one of the plurality of voice terminal devices serves as a master device, and the other voice terminal devices serve as slave devices; the master device initiates an extended broadcast carrying a network identifier corresponding to the respondable broadcast and a respondable broadcast of time slot scheduling information to the slave device; the slave device obtains the network identifier by scanning and analyzing the extended broadcast, and synchronizes to the respondable broadcast based on the network identifier and the master device to intervene in the network; and according to the time slot scheduling information, sending uplink voice data to the master device or receiving downlink voice data from the master device in the allocated time slot so as to perform bidirectional voice communication among the plurality of devices, the respondable broadcast is a Bluetooth low-power-consumption periodic respondable broadcast, so that bidirectional and real-time voice communication among a plurality of devices is realized, the efficiency and flexibility of multi-person voice interaction are remarkably improved, and the low-power-consumption characteristic of the communication process is ensured.

Description

TECHNICAL FIELD

[0001] The present application relates to the technical field of wireless communication, in particular to a multi-to-multi voice communication system, method, electronic device and storage medium. BACKGROUND

[0002] As the most basic and direct information interaction way of human beings, voice communication has long played a key role in both long-distance and near-field scenarios. Traditionally, real-time voice communication mainly relies on two types of infrastructure: one is the operator-based cellular network (such as GSM, LTE, 5G), and the other is the local coverage-based local wireless network (such as Wi-Fi). With the rapid development of the Internet of Things, the Industrial Internet and wearable devices, in specific application scenarios (such as factory workshop team collaboration, warehouse logistics real-time scheduling, outdoor emergency rescue command, multi-player competitive game communication, smart teaching interaction, etc.), the demand for near-field multi-to-multi voice communication systems that can be self-organized, low-power, low-latency and support multiple people talking at the same time is increasingly prominent.

[0003] At present, the mainstream technical solutions for realizing multi-to-multi voice communication and their inherent defects are as follows: 1. Group intercom based on cellular network, which is represented by Push-to-Talk over Cellular. The technical principle is to transmit voice stream media through a public cellular network and forward by a server. This scheme depends on the public network and the operator, and cannot be used in areas without network coverage or weak signal (such as underground garages, remote mountainous areas, and large factory interiors), and generates continuous data flow charges; the latency and stability are affected by the network, and when the network is congested, the voice delay increases significantly and the jitter is severe, affecting the real-time interaction experience; the device power consumption is high, and maintaining a cellular network connection consumes a lot of power, which is not conducive to portable or wearable devices with high endurance requirements; 2. Voice transmission based on traditional Bluetooth, classic Bluetooth uses synchronous connection-oriented links for point-to-point voice transmission, which is widely used in Bluetooth headsets, car hands-free, etc. The connection topology is limited, and it is essentially a "one-to-one" or "one-to-many" master-slave architecture, which cannot realize real and efficient full-duplex voice interaction among multiple devices on a single network. To extend to multi-person call, it needs to rely on complex multi-hop forwarding or external servers, resulting in delay accumulation and topology rigidity; the real-time performance is also insufficient, and the SCO / eSCO link is complex to schedule in multi-point connection, making it difficult to guarantee low delay and voice synchronization in a multi-person speaking scenario; and the power consumption and the number of connections are contradictory, maintaining multiple concurrent connections will cause the power consumption of the master device to rise sharply, and the number of connections has a hard upper limit; 3. Real-time voice intercom based on Wi-Fi, which uses the high bandwidth and multicast capability of Wi-Fi to build a VoIP system in a local area network. The power consumption is too high, the power consumption of the Wi-Fi module is much higher than that of the low-power Bluetooth, which is difficult to apply to portable devices that rely on small batteries; network deployment and interference require pre-deployed access points, which are prone to same-frequency interference and packet loss in device-dense environments; the self-organizing network mode is complex, and the number of device connections is limited by the performance of the access point; mobility and fast networking capability are weak, and the process of device discovery, association and IP address allocation is relatively complex, which is not conducive to fast and temporary group formation.

[0004] The prior art has significant shortcomings in meeting the voice communication requirements of the five key characteristics of near field, self-organization, low power consumption, low latency, and true multi-to-multi. In recent years, the Bluetooth Technology Alliance introduced a periodic broadcast and response mechanism in Bluetooth 5.4 specification. The PAwR mechanism supports devices to send data in a predetermined periodic broadcast window and allows a large number of devices to respond efficiently within a specified sub-event, which provides underlying protocol support for implementing a new, decentralized, and low-power group communication. However, how to design a complete system and method based on this mechanism to carry and optimize real-time voice services and solve the inherent problems of multi-to-multi voice communication has not yet been seen a mature public solution. Therefore, an innovative technical solution is urgently needed to fill this technical gap. SUMMARY

[0005] In view of this, embodiments of the present invention provide a many-to-many voice communication system, method, electronic device and storage medium to solve the problems of limited number of connections, high power consumption, large latency, dependence on infrastructure and inability to realize decentralized self-organizing networks in existing near-field voice communication solutions (such as traditional Bluetooth, Wi-Fi and cellular networks) when supporting real-time full-duplex calls for multiple people.

[0006] One aspect of the present invention provides a many-to-many voice communication system, the system comprising a plurality of voice terminal devices, wherein one of the plurality of voice terminal devices serves as a master device and the other voice terminal devices serve as slave devices; The master device sends an extended broadcast carrying the network identifier corresponding to the responsive broadcast and the responsive broadcast containing time slot scheduling information to the slave device; The slave device obtains the network identifier by scanning and parsing the extended broadcast, and synchronizes with the master device to the responsive broadcast based on the network identifier to join the network; and according to the time slot scheduling information, sends uplink voice data to the master device or receives downlink voice data from the master device within the allocated time slot to conduct bidirectional voice communication between multiple devices; the responsive broadcast is a Bluetooth Low Energy periodic responsive broadcast.

[0007] Another aspect of the present invention provides a many-to-many voice communication method, the method comprising: Send an extended broadcast carrying the network identifier corresponding to the responsive broadcast and the responsive broadcast containing time slot scheduling information to the slave device; Receive uplink voice data from a slave device; process the uplink voice data and transmit it via the responsive broadcast.

[0008] Another aspect of the present invention provides a different many-to-many voice communication method, the method comprising: Scan and parse the extended broadcast initiated by the master device, obtain the network identifier carried in the extended broadcast, and synchronize with the responsive broadcast sent by the master device based on the network identifier to enter the network; Based on the time slot scheduling information in the responsive broadcast, uplink voice data is sent to the master device within the allocated time slot; and voice data is received from the master device via the responsive broadcast.

[0009] Another aspect of the present invention provides an electronic device, comprising: processor; Memory used to store the processor's executable instructions; The processor is configured to execute the instructions to implement the many-to-many voice communication method described in any of the above descriptions.

[0010] Another aspect of the present invention provides a computer-readable storage medium that, when the instructions in the storage medium are executed by a processor of an electronic device, enables the electronic device to perform the many-to-many voice communication method described above.

[0011] The multi-to-multi voice communication system provided by this invention includes multiple voice terminal devices, one of which acts as the master device and the others as slave devices, thus forming a communication network foundation supporting multi-person voice interaction. The master device is used to send extended broadcasts containing network identifiers and responsive broadcasts containing time slot scheduling information. Through the cooperation of extended broadcasts and responsive broadcasts, the network is provided with dual guarantees of efficient discovery and reliable communication, improving the flexibility of network construction and the convenience of device access. Slave devices scan and parse the extended broadcasts to obtain the network identifier and synchronize with the master device based on the identifier to enter the network. Then, according to the time slot scheduling information in the responsive broadcasts, they send uplink voice data to the master device or receive downlink voice data from the master device within the allocated time slots, realizing bidirectional, real-time voice communication between multiple devices, significantly improving the efficiency and flexibility of multi-person voice interaction, and ensuring low power consumption during the communication process. Attached Figure Description

[0012] To more clearly illustrate the technical solutions and advantages in the embodiments of the present invention or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0013] Figure 1 This is a communication flowchart of a many-to-many voice communication system provided in an embodiment of this application; Figure 2 A flowchart illustrating a many-to-many voice communication method provided in an embodiment of this application; Figure 3 A flowchart illustrating another many-to-many voice communication method provided in an embodiment of this application; Figure 4 This is a schematic diagram of the structure of a many-to-many voice communication device according to an exemplary embodiment; Figure 5 This is a schematic diagram of another many-to-many voice communication device provided according to an exemplary embodiment. Detailed Implementation

[0014] To further illustrate the technical means and effects of the present invention in achieving its intended purpose, the following detailed description of the specific implementation methods, structures, features, and effects of the present invention, in conjunction with the accompanying drawings and preferred embodiments, is provided below.

[0015] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0016] It should be noted that the terms "first," "second," etc., in the specification and drawings of this invention are used to distinguish similar objects and are not necessarily used to describe a specific order or sequence. It should be understood that such data can be interchanged where appropriate so that the embodiments of the invention described herein can be implemented in orders other than those illustrated or described herein. Furthermore, the terms "comprising" and "having," and any variations thereof, are intended to cover non-exclusive inclusion; for example, a process, method, system, product, or server that comprises a series of steps or units is not necessarily limited to those steps or units explicitly listed, but may include other steps or units not explicitly listed or inherent to such processes, methods, products, or devices.

[0017] Please see Figure 1 , Figure 1 This is a communication flowchart of a many-to-many voice communication system provided in an embodiment of this application, such as... Figure 1 As shown, specifically, in the above-mentioned many-to-many voice communication system, there are multiple voice terminal devices, one of which serves as the master device and the other voice terminal devices serve as slave devices. The master device sends an extended broadcast carrying the network identifier corresponding to the responsive broadcast and the responsive broadcast containing time slot scheduling information to the slave device; The slave device obtains the network identifier by scanning and parsing the extended broadcast, and synchronizes with the master device to the responsive broadcast based on the network identifier to enter the network; and according to the time slot scheduling information, sends uplink voice data to the master device or receives downlink voice data from the master device within the allocated time slot to conduct bidirectional voice communication between multiple devices.

[0018] In one specific embodiment, the voice terminal device can be a physical device capable of voice input, processing, and output, such as a smart walkie-talkie, Bluetooth headset, industrial head-mounted device, or smartwatch. Specifically, in this system, each voice terminal device has the capability to participate in many-to-many voice communication. Responsive broadcast refers to Bluetooth Low Energy's periodic responsive broadcast. Specifically, responsive broadcast refers to a communication mechanism that supports bidirectional data exchange between Bluetooth Low Energy devices within a periodic broadcast window. Optionally, it is a new broadcast mode defined in the Bluetooth core specification that allows slave devices to respond to broadcast data within a specified time window, providing the underlying time-slotted transmission foundation for many-to-many communication. The network identifier can be a code or information field used to uniquely identify or distinguish a specific voice communication network. Optionally, the network identifier is included in the broadcast data packet initiated by the master device. Slave devices identify this identifier to determine and select which target network to join, ensuring isolation between different communication groups. Time slot scheduling information can be control information generated and issued by the master device, used to specify or allocate specific time segments for each slave device to perform uplink (transmit) or downlink (receive) communication within a broadcast cycle. Specifically, time slot scheduling information defines the timing structure of communication and is key to coordinating the orderly and conflict-free sharing of the wireless channel by multiple devices. Uplink voice data refers to the voice data stream collected, encoded, and prepared for transmission to the master device by the slave device, with the direction from slave device to master device. Downlink voice data refers to the voice data stream processed (e.g., mixed) by the master device and prepared for broadcast to all slave devices in the network, with the direction from master device to slave device. Specifically, during system initialization, one voice terminal device is designated as the master device, and the remaining devices act as slave devices. The core responsibility of the master device is to act as the network coordination center. It initiates a spread broadcast at a low frequency and periodically initiates a special responsive broadcast. The spread broadcast carries a network identifier to announce a specific communication group; the responsive broadcast contains time slot scheduling information to establish a precise schedule for communication activities within the network.

[0019] The slave device continuously performs wireless scanning. When it detects an extended broadcast, it parses it and extracts the network identifier. If the identifier matches the network it wants to join, the slave device synchronizes with the master device to the respondable broadcast based on the time slot scheduling information in the respondable broadcast corresponding to the network identifier, thus officially joining the network.

[0020] Once the network is successfully established, the system enters a stable many-to-many voice communication phase. Each slave device strictly follows the time slot scheduling information issued by the master device: within its assigned uplink time slot, it sends uplink voice data collected from its local microphone and encoded and compressed to the master device; while in other time slots (usually the master device's broadcast time slots), it listens on the channel, receives downlink voice data from the master device (this data may contain the result of the master device mixing uplink voice data from multiple slave devices or other control information), decodes it, and plays it to the user.

[0021] In an optional embodiment, the master device is further configured to receive uplink voice data from a plurality of the slave devices, and to perform mixing or selection processing on the uplink voice data to obtain processed voice data; and to send the processed voice data to the slave devices via the responsive broadcast.

[0022] In one specific embodiment, during each communication cycle, the master device receives uplink voice data packets from multiple slave devices in the network. Subsequently, the audio processing unit within the master device processes these concurrent voice data streams in real time according to the application scenario's requirements. If the scenario needs to simulate a natural group chat atmosphere (such as a team discussion), a mixing algorithm is used to intelligently mix multiple voice signals to generate a single audio stream containing the voices of all active speakers, ensuring that each participant can hear others speaking simultaneously. In scenarios requiring a clear chain of command or to avoid voice conflicts (such as construction site scheduling), a selective processing strategy can be adopted, such as selecting a primary voice stream for forwarding based on "key-to-speak" priority or the strongest signal. Furthermore, after processing, the master device broadcasts the generated processed voice data (i.e., the mixed audio stream or the selected primary voice stream) as part of its periodically responsive broadcast payload to all slave devices in the network.

[0023] In the above embodiments, a closed loop of "uplink aggregation, central processing, and unified broadcast downlink" for voice data is realized. In this way, the system not only solves the conflict problem of concurrent transmission of multiple voice channels, but also greatly optimizes the network bandwidth utilization and ensures the consistency of the content listened to by all participants. Thus, under the limited resource constraints of Bluetooth Low Energy, a stable, clear and natural multi-person real-time voice call experience is achieved.

[0024] In an optional embodiment, before sending the uplink voice data, the slave device needs to request and obtain the uplink time slot allocated by the master device through the time slot scheduling information.

[0025] In one specific embodiment, when a slave device needs to transmit voice, it cannot arbitrarily occupy the channel but must first initiate an uplink time slot request to the master device. This request is typically sent within a control time slot specifically reserved for this purpose, or it is accomplished via signaling carried in a response data packet. The master device, acting as the centralized controller of the network, continuously collects all transmission requests from slave devices. Subsequently, the master device runs a scheduling algorithm (such as first-come, first-served, round-robin, or priority-based scheduling) and dynamically generates or updates the time slot scheduling information to be broadcast in the next cycle. This scheduling information explicitly indicates which slave device has transmission permission in which specific sub-event (uplink time slot). The master device publicizes this authorization information to the entire network through its periodic responsive broadcasts; optionally, the slave device that has requested transmission permission can determine whether it has obtained an uplink time slot and its specific time slot location by listening to and parsing this broadcast. Only when a slave device explicitly "obtains" a specific uplink time slot allocated by the master device is it allowed to initiate the transmission of its uplink voice data at that precise moment.

[0026] In the above embodiments, the slave device ensures conflict-free access to the uplink channel through a request-before-send mechanism. This design avoids packet collisions and retransmission overhead caused by random contention among multiple devices, significantly reducing communication latency and power consumption. Simultaneously, the master device can dynamically allocate time slots based on real-time network load and priority, optimizing bandwidth utilization. This allows the system to maintain high reliability and low latency even when supporting more concurrent voice streams, making it suitable for voice application scenarios with stringent real-time requirements.

[0027] In an optional embodiment, the above-mentioned voice terminal device includes: Bluetooth control unit, used to support and execute the Bluetooth periodic broadcast and response protocol; An audio acquisition unit is used to acquire the user's voice signal; An audio encoding / decoding unit is used to encode and compress the speech signal and to decode the received encoded speech data. The audio playback unit is used to play the decoded audio signal.

[0028] In one specific embodiment, the Bluetooth Control Unit (BCU) is the "brain" and "transceiver" of the device for wireless communication. Specifically, its core function is to support and execute the Bluetooth periodic broadcast and response protocol. Optionally, when the device acts as a master device, this module is responsible for generating and periodically sending broadcast data packets containing network identifiers and time slot scheduling information; when the device acts as a slave device, this module is responsible for scanning channels, parsing master device broadcasts, and sending response data packets or receiving broadcast data within precisely allocated time slots. This module implements the underlying PAWR mechanism, responsible for precise timing synchronization, frequency hopping, and radio frequency control, and is the cornerstone of the entire communication link.

[0029] Audio acquisition units typically include microphones and preamplifiers, which are responsible for converting the user's physical voice signal into an analog electrical signal, and then converting it into raw digital audio data through an analog-to-digital converter.

[0030] The audio codec unit is a key component for processing voice data and achieving efficient transmission. It performs bidirectional processing: first, encoding (compression) processing, specifically encoding and compressing the high-quality, high-bandwidth raw digital audio data sent by the audio acquisition unit, using low-complexity, low-latency codecs such as LC3 to significantly reduce the amount of data that needs to be transmitted wirelessly, in order to adapt to the limited bandwidth of Bluetooth Low Energy; second, decoding processing, specifically decoding the compressed encoded voice data received through the Bluetooth control unit to restore it to a playable audio data format.

[0031] An audio playback unit typically includes an audio amplifier, a speaker, or a headphone jack. It is responsible for converting the digital audio signal output from the decoding unit into an analog signal and driving the speaker to produce sound, ultimately restoring the decoded voice signal into a sound that the user can hear.

[0032] Specifically, each voice terminal device is an intelligent terminal integrating wireless communication, audio processing, and user interaction functions. Its hardware architecture revolves around four core modules that work together to achieve many-to-many voice communication. The Bluetooth control unit is the absolute core of the device's interaction with the wireless world. It is not merely a traditional Bluetooth chip, but is specifically optimized or configured to support and execute the Bluetooth periodic broadcast and response protocol. In master device mode, this module acts as a "network coordinator" and "broadcast tower," strictly sending scheduling information according to the cycle; in slave device mode, it transforms into a "precise listener," strictly completing data transmission or reception within the allocated micro-slots, thus achieving conflict-free, low-latency time-slotted communication between multiple devices at the physical layer. The audio acquisition unit, encoding / decoding unit, and playback unit together constitute the device's "voice processing pipeline." The audio acquisition unit, like the device's "ears," is responsible for picking up user voice and completing initial digitization. The audio encoding / decoding unit is the "translator" and "compression engineer," efficiently compressing uplink voice to save valuable wireless bandwidth and decompressing downlink voice to restore sound. Finally, the audio playback unit acts as the "mouth," clearly playing the processed sound to the user.

[0033] In the above embodiment, the four modules are tightly coupled through system software and an internal bus: the acquired voice is encoded and sent by the Bluetooth module; the data received by the Bluetooth module is decoded and output by the playback unit. This modular design ensures that the device can flexibly play the role of master or slave, and can efficiently complete the entire voice communication process, thereby supporting a stable, clear, and scalable many-to-many voice communication network at the system level.

[0034] In an optional embodiment, the roles of the master device and the slave device can be switched according to preset rules or dynamic negotiation.

[0035] In one specific embodiment, preset rule switching can be a system that automatically triggers role switching based on pre-defined and explicit conditions. Optionally, preset rule switching can include power priority, signal strength priority, device capability priority, and manual specification. Dynamic negotiation switching allows devices in the network to interact with specific control signaling, jointly decide and complete role switching based on real-time conditions during runtime. Specifically, the role of a device in the network (master device or slave device) is not fixed, but can be switched according to preset rules or through dynamic negotiation. For example, the system can preset a rule: when the remaining power of the master device is less than 20%, the switching process is automatically triggered, transferring the master device's responsibilities to the slave device with the most abundant power in the network, thereby avoiding the paralysis of the entire communication network due to the master device shutting down. As another example, in a mobile scenario, if the current master device gradually moves away from the group center, its connection quality with edge slave devices deteriorates. The system can use dynamic negotiation (such as election based on mutual signal strength reports between devices) to seamlessly switch the master device's role to a device with a more central geographical location or better signal conditions to maintain optimal network connection quality.

[0036] In the above embodiments, the dynamic role switching mechanism significantly enhances the robustness and adaptability of the system. By automatically switching the master device based on battery level, signal strength, or negotiation results, it avoids network paralysis caused by single points of failure and improves the system's continuous service capability in mobile scenarios or when device states change. Simultaneously, it optimizes network topology and resource allocation, ensuring low latency and high stability for voice communication, realizing a truly decentralized self-organizing network, and significantly improving user experience and system reliability.

[0037] In an optional embodiment, the aforementioned voice terminal device is further configured to perform audio repair by predictive compensation or frame interpolation methods when packet loss of received voice data is detected.

[0038] In one specific embodiment, voice data may lose some data packets during wireless transmission due to interference, collisions, or signal attenuation. Packet loss audio repair aims to mitigate the impact of packet loss on the listening experience. Specifically, when the device's audio processing unit detects packet loss in the received encoded voice data stream (e.g., through discontinuous packet sequence numbers or checksum errors), the repair mechanism is immediately activated. Instead of simply filling the gap with silence or abruptly repeating the previous frame (which would produce a noticeable "click" or interruption), the device employs a more intelligent algorithm for repair. One primary method is frame interpolation: the algorithm acquires successfully decoded audio frames before and after the lost data packets, analyzes their acoustic features such as spectrum and pitch, and generates a smooth, naturally connected audio signal through interpolation to fill the gap caused by packet loss. Another method is predictive compensation, for example, based on the principle of linear predictive coding, using the voice signal before packet loss to establish a short-term predictive model, and using this model to generate a signal that continues the previous voice characteristics as compensation. This repair process is completed locally and in real-time on the device, without requiring retransmission from the master device or additional network signaling, thus significantly masking the impact of packet loss on call quality with minimal additional latency.

[0039] In the above embodiments, the packet loss audio repair mechanism effectively improves the anti-interference capability and subjective listening quality of voice communication. By intelligently repairing lost voice data packets on the terminal side, it significantly reduces voice stuttering, interruptions, or noise caused by unstable wireless channels, making calls sound more coherent and natural.

[0040] The following describes a many-to-many voice communication method provided by an embodiment of this application. This method is implemented based on the aforementioned device with many-to-many voice communication functionality. Figure 2 This is a flowchart illustrating a many-to-many voice communication method provided in an embodiment of this application. It should be noted that this specification provides the operational steps of the method described in the embodiments or flowcharts, but based on conventional or non-inventive labor, more or fewer operational steps may be included. The order of steps listed in the embodiments is merely one possible execution order among many steps and does not represent the only execution order. In actual systems or products, the methods can be executed sequentially or in parallel (e.g., in a parallel processor or multi-threaded processing environment) as shown in the embodiments or accompanying drawings. Specifically, as... Figure 2 As shown, the above method may include: 201: Send an extended broadcast carrying the network identifier corresponding to the responsive broadcast and the responsive broadcast containing time slot scheduling information to the slave device; S203: Receive uplink voice data from the slave device; process the uplink voice data and transmit it via a responsive broadcast.

[0041] In a specific embodiment, the many-to-many voice communication method defined in this invention is manifested on the master device side as a periodic cyclical operation of "scheduling, aggregation, and broadcasting". The method begins with the master device acting as the network timing and coordination center, continuously initiating extended broadcasts and periodically initiating responsive broadcasts. The extended broadcasts carry the network identifier of the responsive broadcasts, and the responsive broadcasts embed time slot scheduling information. Based on the uplink time slot specified by the time slot scheduling information, the master device switches to receiving mode and receives uplink voice data from slave devices. This data represents the real-time voice input of each participant in the network. Subsequently, the core value of the master device is realized: it processes the received multiple voice streams in real time. In typical group chat scenarios, the master device performs mixing processing, intelligently fusing multiple voice streams to generate a single mixed audio stream simulating a real multi-person dialogue sound field; in command and dispatch scenarios, it may perform selection processing, forwarding only the highest priority voice stream. After processing, the master device sends the generated integrated voice data as the payload for its next round of periodic responsive broadcasts. Through this closed-loop process, the master device successfully aggregates, processes, and distributes the dispersed and concurrent uplink voice streams to all participants, thereby efficiently and stably realizing the logical function of many-to-many full-duplex voice communication with the limited resources of Bluetooth Low Energy.

[0042] The following describes another many-to-many voice communication method provided in the embodiments of this application. This method is also based on the aforementioned device with many-to-many voice communication function. Figure 3 This is a flowchart illustrating a many-to-many voice communication method provided in an embodiment of this application. It should be noted that this specification provides the operational steps of the method described in the embodiments or flowcharts, but based on conventional or non-inventive labor, more or fewer operational steps may be included. The order of steps listed in the embodiments is merely one possible execution order among many steps and does not represent the only execution order. In actual systems or products, the methods can be executed sequentially or in parallel (e.g., in a parallel processor or multi-threaded processing environment) as shown in the embodiments or accompanying drawings. Specifically, as... Figure 3 As shown, the above method may include: S301: Scan and parse the extended broadcast initiated by the master device, obtain the network identifier carried in the extended broadcast, and synchronize with the responsive broadcast sent by the master device based on the network identifier to intervene in the network; S303: Based on the time slot scheduling information in the responsive broadcast, send uplink voice data to the master device within the allocated time slot; and receive voice data sent from the master device via the responsive broadcast.

[0043] In one specific embodiment, the many-to-many voice communication method defined in this invention is embodied on the slave device side as a scheduled operation mode of "discovery-synchronization-controlled transmission-continuous reception". The method begins with the slave device actively scanning. When it captures an extended broadcast from the master device, it immediately parses it, extracts key network identifiers to confirm the target group, and obtains reference information for time synchronization. Subsequently, the device uses this information to precisely synchronize its internal clock with the master device's core communication mechanism—the period that responds to broadcasts. Once synchronization is successful, the device officially joins the self-organizing network, transforming into a scheduled slave device.

[0044] Once in steady-state operation, all communication behaviors of the slave devices strictly adhere to the centralized scheduling of the master device. It continuously parses the time slot scheduling information carried in each broadcast cycle, which acts like a "communication timetable." According to this table, the slave device is only authorized to initiate a transmission link and transmit its uplink voice data to the master device within its explicitly allocated time slots, thus avoiding wireless conflicts between multiple devices. Simultaneously, for the majority of its operating time, the slave device is in a receiving state, continuously receiving voice data from the master device via responsive broadcasts. This downlink data includes group call content integrated by the master device. Through this combination of strictly controlled transmission and continuous reception, each slave device can both contribute its own voice and synchronously perceive the voice status of the entire network, thereby achieving the core function of participating in high-quality, real-time many-to-many voice communication as a terminal under low power consumption and low complexity conditions.

[0045] This invention provides a many-to-many voice communication system, method, electronic device, and storage medium, which forms a self-organizing communication network by designating one device as the master device and the rest as slave devices among multiple voice terminal devices. On the master device side, a Bluetooth Low Energy periodic responsive broadcast containing network identifier and time slot scheduling information is periodically initiated. On the slave device side, the slave device scans and parses the broadcast to synchronously access the network, and sends uplink voice data or receives downlink voice data sent by the master device within the allocated time slot according to the time slot scheduling information, realizing bidirectional voice communication between multiple devices. Furthermore, the master device receives uplink voice data from multiple slave devices, performs mixing or selection processing, and then sends the processed voice data to all slave devices through responsive broadcasts, effectively integrating multiple voice streams. Furthermore, the voice terminal device integrates a Bluetooth control unit supporting the PAWR protocol, audio acquisition, encoding / decoding, and playback units, providing a hardware foundation for low-power, high real-time voice processing. Furthermore, by designing a mechanism that requires slave devices to request and obtain uplink time slots before sending, channel conflicts are avoided. By introducing dynamic switching of device roles and packet loss audio repair functions, network robustness and call clarity are enhanced, respectively. This solution is based on the Bluetooth PAwR mechanism and constructs a decentralized, self-organizing near-field many-to-many real-time voice communication system, which significantly improves the real-time performance, stability and flexibility of multi-person voice interaction, while maintaining the system's low power consumption characteristics. It is suitable for various scenarios such as industrial collaboration, emergency command, and team communication.

[0046] Figure 4 This is a schematic diagram of a many-to-many voice communication device according to an exemplary embodiment. The device is implemented based on the many-to-many voice communication system described above. Figure 4 As shown, the above-mentioned device includes: The broadcast initiation module is used to send an extended broadcast carrying the network identifier corresponding to the responsive broadcast and the responsive broadcast containing time slot scheduling information to the slave device; An uplink processing module is used to receive uplink voice data from a slave device; process the uplink voice data; and transmit it via the responsive broadcast.

[0047] Figure 5 This is a schematic diagram of another many-to-many voice communication device according to an exemplary embodiment. The device is implemented based on the many-to-many voice communication system described above. Figure 5 As shown, the above-mentioned device includes: The broadcast parsing module is used to scan and parse extended broadcasts initiated by the master device, obtain the network identifier carried in the extended broadcast, and synchronize with the responsive broadcast sent by the master device based on the network identifier to enter the network; The downlink receiving module is configured to send uplink voice data to the master device within the allocated time slot according to the time slot scheduling information in the responsive broadcast; and to receive voice data sent by the master device through the responsive broadcast.

[0048] In an exemplary embodiment, an electronic device is also provided, including: a processor; and a memory for storing processor-executable instructions; wherein the processor is configured to execute the instructions to implement a many-to-many voice communication method as described in the embodiments of this disclosure.

[0049] In an exemplary embodiment, a computer-readable storage medium is also provided, which, when the instructions in the storage medium are executed by a processor of an electronic device, enables the electronic device to perform the many-to-many voice communication method of the present disclosure embodiments.

[0050] In an exemplary embodiment, a computer program product or computer program is also provided, which includes computer instructions stored in a computer-readable storage medium. A processor of a computer device reads the computer instructions from the computer-readable storage medium and executes the computer instructions, causing the computer device to perform the many-to-many voice communication method provided in the various optional implementations described above.

[0051] Those skilled in the art will understand that all or part of the processes in the methods of the above embodiments can be implemented by a computer program instructing related hardware. This computer program can be stored in a non-volatile computer-readable storage medium. When executed, the computer program can include the processes of the embodiments of the above methods. Any references to memory, storage, databases, or other media used in the embodiments provided by this invention can include non-volatile and / or volatile memory. Non-volatile memory can include read-only memory (ROM), programmable ROM (PROM), electrically programmable ROM (EPROM), electrically erasable programmable ROM (EEPROM), or flash memory. Volatile memory can include random access memory (RAM) or external cache memory. By way of illustration and not limitation, RAM is available in various forms, such as static RAM (SRAM), dynamic RAM (DRAM), synchronous DRAM (SDRAM), dual data rate SDRAM (DDRSDRAM), enhanced SDRAM (ESDRAM), synchronous link DRAM (SLDRAM), RAMbus direct RAM (RDRAM), direct memory bus dynamic RAM (DRDRAM), and RAMbus dynamic RAM (RDRAM), etc.

[0052] Other embodiments of this disclosure will readily occur to those skilled in the art upon consideration of the specification and practice of the invention disclosed herein. This invention is intended to cover any variations, uses, or adaptations of this disclosure that follow the general principles of this disclosure and include common knowledge or customary techniques in the art not disclosed herein. The specification and examples are to be considered exemplary only, and the true scope and spirit of this disclosure are indicated by the foregoing claims.

[0053] It should be understood that this disclosure is not limited to the precise structures described above and shown in the accompanying drawings, and various modifications and changes can be made without departing from its scope. The scope of this disclosure is limited only by the appended claims. The above description is merely a preferred embodiment of the present invention and is not intended to limit the present invention in any way. Although the present invention has been disclosed above with reference to preferred embodiments, it is not intended to limit the present invention. Any person skilled in the art can make some modifications or alterations to the above-disclosed technical content to create equivalent embodiments without departing from the scope of the present invention. Any simple modifications, equivalent changes and alterations made to the above embodiments based on the technical essence of the present invention without departing from the scope of the present invention shall still fall within the scope of the present invention.

Claims

1. A many-to-many voice communication system, characterized in that, It includes multiple voice terminal devices, one of which serves as the master device and the other voice terminal devices serve as slave devices; The master device sends an extended broadcast carrying the network identifier corresponding to the responsive broadcast and the responsive broadcast containing time slot scheduling information to the slave device; The slave device obtains the network identifier by scanning and parsing the extended broadcast, and synchronizes with the master device to the responsive broadcast based on the network identifier to join the network; and according to the time slot scheduling information, sends uplink voice data to the master device or receives downlink voice data from the master device within the allocated time slot to conduct bidirectional voice communication between multiple devices; the responsive broadcast is a Bluetooth Low Energy periodic responsive broadcast.

2. The system according to claim 1, characterized in that, The master device is also configured to receive uplink voice data from multiple slave devices, and to perform mixing or selection processing on the uplink voice data to obtain processed voice data; and to send the processed voice data to the slave devices via the responsive broadcast.

3. The system according to claim 2, characterized in that, The voice terminal device includes: Bluetooth control unit, used to support and execute the Bluetooth periodic broadcast and response protocol; An audio acquisition unit is used to acquire the user's voice signal; An audio encoding / decoding unit is used to encode and compress the speech signal and to decode the received encoded speech data. The audio playback unit is used to play the decoded audio signal.

4. The system according to claim 1, characterized in that, The roles of the master device and the slave device can be switched according to preset rules or dynamic negotiation.

5. The system according to claim 1, characterized in that, Before sending the uplink voice data, the slave device needs to request and obtain the uplink time slot allocated by the master device through the time slot scheduling information.

6. The system according to claim 1, characterized in that, The voice terminal device is also used to repair audio by predictive compensation or interpolation of previous and next frames when packet loss of received voice data is detected.

7. A many-to-many voice communication method, applied to the system as described in any one of claims 1-6, characterized in that, The method includes: Send an extended broadcast carrying the network identifier corresponding to the responsive broadcast and the responsive broadcast containing time slot scheduling information to the slave device; Receive uplink voice data from a slave device; process the uplink voice data and transmit it via the responsive broadcast.

8. A many-to-many voice communication method, applied to the system as described in any one of claims 1-6, characterized in that, The method includes: Scan and parse the extended broadcast initiated by the master device, obtain the network identifier carried in the extended broadcast, and synchronize with the responsive broadcast sent by the master device based on the network identifier to enter the network; Based on the time slot scheduling information in the responsive broadcast, uplink voice data is sent to the master device within the allocated time slot; and voice data is received from the master device via the responsive broadcast.

9. An electronic device, characterized in that, include: processor; Memory used to store the processor's executable instructions; The processor is configured to execute the instructions to implement the many-to-many voice communication method as described in any one of claims 7 to 8.

10. A computer-readable storage medium, characterized in that, When the instructions in the storage medium are executed by the processor of the electronic device, the electronic device is able to perform the many-to-many voice communication method as described in any one of claims 7 to 8.

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