A communication assistance method, master device, system and storage medium

Abstract

The application discloses a communication auxiliary system, a master control device and a method, and belongs to the technical field of communication. The system adopts a mediation double-agent architecture, the master control device is logically connected between a communication terminal and a headset device, and meanwhile, isolated physical connections are established by simulating hands-free (HF) device specifications and audio gateway (AG) device specifications. The master control device is internally provided with an uplink data decision module and a memory bus with a first buffer address and a backup buffer address. When a local trigger event (such as touch, action or external entity button pulse) is captured, the decision module forcibly switches the memory data reading pointer of the digital audio by using the bottom-layer hardware atomic operation in an instant, and replaces the headset pickup source with an external high-definition pickup source. The whole switching process does not trigger the bottom-layer audio routing redistribution (Audio Policy) of the communication terminal operating system, and the absolute continuity of the Bluetooth bottom-layer SCO synchronous link frame sequence is maintained. The system realizes the high-response seamless migration and injection of the external pickup source without invading the protocol stack of the host device, supports the double-track solidification of the multi-dimensional voice shunting of the communication link, and significantly improves the communication and pickup efficiency in modern mobile and complex scenarios.

Description

Technical Field

[0001] This application relates to the field of communication technology, specifically to a communication assistance method, main control device, system, and readable storage medium in a multi-terminal collaborative scenario. Background Technology

[0002] With the rapid development of wireless communication technology and smart wearable devices, wireless headphones (such as TWS (True Wireless Stereo) headphones, neckband Bluetooth headphones, etc.) have become standard peripheral devices for various communication terminals (usually referred to as host devices) such as smartphones and tablets.

[0003] In traditional wireless audio interaction architectures, a "point-to-point" star topology is typically used: wireless headphones establish a direct connection with the mobile phone, which acts as the communication hub, via a single short-range wireless communication protocol (such as the HFP (Hands-Free Profile) in the Bluetooth protocol stack). Under this architecture, the mobile phone's operating system logically monopolizes gateway control over both uplink (sound pickup) and downlink (sound playback) data streams.

[0004] This single-chain architecture, which is highly dependent on the host device, reveals significant limitations when dealing with complex high-fidelity communication or high-frequency concurrent processing scenarios.

[0005] Firstly, there is the strong coupling and high latency of audio routing switching. When users are in noisy environments or special duty scenarios and expect to seamlessly switch the audio source from the currently attached wireless headset to a lavalier high-definition microphone or an external device with a powerful noise reduction physical matrix during a call, the system must initiate audio routing reallocation at the protocol layer from the mobile phone (e.g., triggering the `Audio Policy Manager` event mechanism). This process involves a re-handshake of higher-level drivers, inevitably leading to audio disconnection and popping at the millisecond or even second level.

[0006] Secondly, there is a lack of physical data aliasing and source-end isolation capabilities. In the existing communication recording system, whether it is call recording or data sent to the cloud for AI voice processing, it all comes from the audio stream after the mobile phone's mixer. Because the downlink far-end sound and the uplink near-end sound are aliased by software before being transmitted to the codec, even with computationally intensive blind source separation (BSS) algorithms or speaker diarization strategies, it is difficult to accurately decouple the overlapping speech caused by simultaneous speaking (intercepting each other) in the post-processing stage.

[0007] Therefore, how to achieve seamless hot-switching of multi-dimensional audio sources on the communication link without intruding on or modifying the underlying protocol stack of the communication terminal, and how to achieve pure multi-track digital voice extraction with absolute isolation at the physical layer, has become a technical problem that urgently needs to be solved by those skilled in the art. Summary of the Invention

[0008] This application provides a communication assistance method, a master control device, a system, and a storage medium, aiming to break through the centralized control of audio uplink and downlink by traditional communication terminals. By constructing an isolated middleware proxy architecture between devices, it achieves efficient autonomous scheduling of audio sources and solidification of communication anti-interference data.

[0009] Objective of the Invention: Firstly, this application aims to provide a "dual-herringbone proxy architecture (or dual-proxy architecture)" that adds an independently controlled proxy layer to the communication link. This proxy layer disguises itself as a hands-free terminal to the upper level and as a calling gateway to the lower level, thereby seizing absolute control over the communication media stream without increasing the computational burden on the host device or calling any unauthorized APIs of the host device.

[0010] Secondly, this application aims to provide a data frame anti-disconnection injection scheme based on the above-mentioned proxy architecture, which solves the network disconnection problem caused by hot-plugging of microphones between multiple devices by swapping data at the underlying memory pointer level.

[0011] Technical solution: In order to achieve the above objectives, the first aspect of this application provides a communication assistance system based on a mediator dual-agent architecture (dual herringbone topology architecture), the system comprising: a first communication node (communication terminal), a master control device, and at least one second communication node (earphone device).

[0012] The main control device in the communication system is configured to: simultaneously simulate the hands-free (HF) device specification to establish a first physical connection with the first communication node, and simultaneously simulate the audio gateway (AG) device specification to establish a second physical connection with the second communication node; that is, the main control device is logically connected in series between the communication terminal and the headset device.

[0013] The main control device is equipped with [equipment / function].

[0014] The baseband concurrent processing module is used to synchronously process and transmit digital audio messages while maintaining the stability of the first physical connection and the second physical connection.

[0015] An uplink data decision module is configured to respond to local triggering events in an autonomous control state; the autonomous control state indicates that the decision module executes switching logic decisions independently of the control commands issued by the first communication node.

[0016] Based on the response of the decision module, an internal data bus performs an operation to forcibly switch the data read pointer of the currently assembled uplink audio packet from the first buffer address to the second buffer address without sending a link suspension or renegotiation signal to the first physical connection.

[0017] The first buffer address temporarily holds an audio feature stream from the second communication node, and the second buffer address temporarily holds an audio feature stream from the local acquisition unit of the main control device.

[0018] Furthermore, as an embodiment of the second aspect of this application, a supplementary feature is provided.

[0019] The baseband concurrent processing module includes an asynchronous sampling rate compensation unit (ASRC). The compensation unit is configured to: extract the target clock frequency of the first physical connection as a reference time base; perform bit-depth adaptive hardware resampling and frame header alignment on the first baseband data sequence originating from the second communication node and the second baseband data sequence originating from the local acquisition unit to address the time delay drift caused by the difference in the anomalous crystal oscillator; and ensure smooth splicing of the data sequence in the time domain during the aforementioned pointer switching operation.

[0020] Beneficial effects: The beneficial effects of the technical solutions provided in the embodiments of this application include at least the following.

[0021] First, by introducing a master control device with a specific structure to form a "double-V" logical link, the uplink and downlink audio permissions are physically separated from the communication terminal (such as a smartphone). This independent control layer blocks possible eavesdropping or unauthorized takeover attempts at the mobile phone software level, forming a strong security barrier in the architectural design.

[0022] Secondly, by using the instantaneous transfer of the underlying data extraction pointer to replace the traditional protocol handshake switching, true zero-latency hot switching is achieved. No matter how frequently the user switches between the external pickup array and the local headphone microphone, the communication terminal is not only "unaware" of the switch, but also avoids any application-level packet loss issues.

[0023] Third, this "dual-agent bridging" inherently provides the ability to intercept baseband audio before it has undergone mixing and digital-to-analog conversion. Coupled with a clock compensation strategy, it lays a decisive hardware foundation for delivering 100% absolutely clean two-track physically isolated data to subsequent intelligent computing or evidence storage terminals. Attached Figure Description

[0024] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the accompanying drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, the accompanying drawings described below are some embodiments of this application. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0025] Figure 1 This is a schematic diagram of the overall communication topology of a communication auxiliary system based on a mediator dual-proxy architecture, provided in an embodiment of this application.

[0026] Figure 2 This application provides a structural block diagram of the core hardware functional modules and audio data flow of a master control device (intermediary hub) in an embodiment of the present application.

[0027] Figure 3 The timing / flowchart of a low-level audio read pointer switching method that does not induce topology reconnection is provided in an embodiment of this application. Detailed Implementation

[0028] To make the objectives, technical solutions, and advantages of this application clearer, the embodiments of this application will be further described below with reference to the accompanying drawings.

[0029] The terminology used in this application is for the purpose of describing particular embodiments only and is not intended to limit the application. The terms "first," "second," etc., in the specification, claims, and accompanying drawings are used to distinguish different objects and not to describe a particular order. Furthermore, the terms "including" and "having," and any variations thereof, are intended to cover non-exclusive inclusion. For example, a process, method, system, product, or apparatus that includes a series of steps or units is not limited to the listed steps or units, but may optionally include steps or units not listed, or may optionally include other steps or units inherent to such processes, methods, products, or apparatus.

[0030] In this document, the terms "master device," "agent device," or "independent intermediate hardware" can refer to physical processing devices that employ microcontrollers (MCUs), digital signal processors (DSPs), application system-on-a-chip (SoCs), or other entities capable of executing embedded firmware and logic instructions. These devices possess their own register control network and independent, directly accessible memory bus interfaces (e.g., DMA controllers).

[0031] The communication terminal mentioned in this document may refer to a smartphone, tablet computer, personal digital assistant, vehicle communication control board, private network walkie-talkie motherboard, or any other initiating / receiving host device with WAN / LAN voice communication capabilities and capable of sending signaling.

[0032] The headphone devices mentioned in this document may include single-ear or dual-ear wireless headphones (such as TWS headphones), over-ear communication headsets, wired electret microphone arrays, or other terminal execution components that can receive electrical signals and convert them into acoustic signals, and have the ability to pick up and provide feedback on local physical environment. Example 1: Overall physical topology and baseband link construction of the system.

[0033] Please refer to Figure 1. This embodiment provides a highly reliable dual-herringbone (or dual-layer cascaded proxy architecture) communication assistance system. In this embodiment, the communication terminal acts as the initiator of the entire external communication link and the upstream source of the voice signal; the headset device acts as the execution end for the user to acquire sound and initially provide feedback on ambient sound; and the main control device acts as the necessary physical connector between the two.

[0034] Specifically.

[0035] The network layer configuration of the master control device contains two independent protocol stack running engines or task threads.

[0036] The first protocol stack engine is configured to present itself to the communication terminal. During the pairing and bonding phase of a specific Bluetooth or other short-range local transmission, the master device actively announces its support for standard hands-free / headset role agreements such as HFP (Hands-Free Profile) or HSP (Headset Profile). This means that, from the perspective of the communication terminal's (e.g., mobile phone) operating system, the master device is equivalent to a regular wireless microphone / headset combination. The communication terminal will then routinely push downlink voice baseband streams (such as downlink SCO / eSCO link data) into the master device's downlink receive buffer through this first physical connection, and be ready at any time to extract voice payload packets from the master device's uplink transmit buffer and transmit them to the remote network.

[0037] The second protocol stack engine is configured to present a host role to the lower layer (headphone device) during this process. In this subnet, the master device transforms into an audio gateway (AG), accepting passive connection requests from the headphone device (terminal recipient). The master device is responsible for distributing synchronization beats for this second physical connection, controlling the connection period, and receiving the first uplink audio sequence submitted in this stage.

[0038] Thus, thanks to this physical dual-link design, the original direct connection between the communication terminal and the headset was physically severed. Without requiring any modification to the existing operating system or upper-layer application drivers on the communication terminal, the scheduling and merging of the system's underlying audio data was forcibly taken over by the main control device. This provides a vacuum isolation layer for subsequent implementation of high-definition audio source hot-switching unaffected by the host and extraction that prevents aliasing and crosstalk. Example 2: Structure and Functional Logic of the Core Agent Module of the Main Control Device

[0039] Based on the physical topology established above, the built-in processing unit of the main control device is further described in detail. Please refer to the main control device structure logic block diagram shown in Figure 2. In this embodiment, the firmware of the main control device is divided into several core sub-modules that work together.

[0040] The first component is the concurrent network stack processing module. Traditional Bluetooth or communication chips, due to memory or instruction limitations, can typically only handle single-role link maintenance (e.g., simply receiving audio as a headset). However, under this patent framework, the concurrent network stack processing module can time-division multiplex on the same baseband processor or through multi-core physical isolation to simultaneously run and maintain the aforementioned first physical connection (HF role) and second physical connection (AG role), with their link encryption and handshake heartbeats operating independently and without interference. This ensures that if the downstream headset device experiences an unexpected power outage, the telephone link connecting the upstream device to the communication terminal (mobile phone) will not be disrupted.

[0041] Secondly, there is the underlying routing controller and the uplink data decision module. This is the key point where this invention breaks through the absolute control of the host mobile phone by the primary device. The main control device is equipped with a local dedicated triggering device (such as a configurable interrupt pin-mapped physical button, a voice-specific wake-word detection array, etc.). The uplink data decision module is in an interrupt monitoring state with the highest preemptive priority. In its decision tree logic branch, physical-level one-way firewall isolation or fake handshake response is set for signals from the communication terminal attempting to take over the microphone or issuing a microphone disable signal (such as some plug-ins that force mute based on private protocols). That is, when the communication terminal issues a control message requesting the cessation of audio acquisition, the decision module replies that it has successfully stopped, but at the physical layer, it still continues to acquire or forward the audio stream according to its own state machine, thereby achieving the system's anti-tampering and strong autonomy. Example 3: Seamless data pointer switching (anti-reconnection strategy) and dynamic injection method

[0042] Furthermore, as a core embodiment of the present invention, this case proposes and details a method for intercepting and replacing uplink audio packets at the underlying level.

[0043] In existing technologies (such as ordinary Bluetooth headsets with external microphones), when a user switches from a built-in microphone to an external microphone, the headset system typically sends a state reset command to the mobile phone according to the Bluetooth Special Interest Group (SIG) HFP specification. After the mobile phone responds, it re-establishes an audio tunnel based on the new endpoint. This process inevitably causes the network link to enter a suspend or even reconnect period.

[0044] However, the firmware of the main control device in this system contains an uplink data decision module and a direct memory access (DMA) or equivalent bus access unit, which executes a technique known as pointer spoofing. Its execution steps include...

[0045] (Step 301) Under normal circumstances, the main control device maintains the first physical connection to the mobile phone, and a set of transmission buffers cyclically extracts baseband packets containing audio payloads and packages them for transmission.

[0046] (Step 302) At this time, the data filling source of the first buffer is locked by a pointer set by DMA to the audio receiving area of ​​the lower protocol stack (i.e., the first uplink voice source from the communication headset).

[0047] (Step 303) The decision module detects a valid logic switching trigger (such as an interrupt or threshold jump).

[0048] (Step 304) The processing unit refuses to report this event. Instead, during the critical gap (usually on the microsecond level) of the next baseband frame to be filled, it performs an atomic operation to directly rewrite the source address pointer of the DMA and forcibly move it to the local acoustic codec buffer (i.e., the second uplink voice source from the local master control large diaphragm pickup array).

[0049] (Step 305) The communication terminal (mobile phone) continues to extract the assembled baseband packets. For the network stack on the mobile phone, the data packets sent are still continuous and unchanged in terms of protocol identifiers such as physical channel address and link packet sequence number. However, the specific audio payload carried by them has been replaced without being noticed.

[0050] This method of forcibly swapping read and write pointers in the memory area completely solves the problem of network disconnection caused by changes in audio routing, making the switching process felt by the user extremely smooth and eliminating disconnections during hot-swapping. Example 4: Clock Domain Forced Compensation and Adaptive Interpolation Reordering Algorithm

[0051] In the seamless handover process of Example 3, since the first uplink voice source (remote wireless headset) and the second uplink voice source (external high-definition microphone or internal pickup array of the main control device) are very likely driven by completely independent physical clock crystal oscillators, there will inevitably be a microscopic deviation in their sampling rates (frequency drift). If memory stream docking is performed directly without intervention, audible periodic clicking sounds (Click / PopNoise) or buffer overflow phenomena will inevitably occur at the receiving end.

[0052] To address this issue, the baseband concurrent processing module in this embodiment includes an advanced asynchronous sampling rate compensation unit (ASRC).

[0053] Within the tolerance window before and after the above switching action is performed.

[0054] 1. The ASRC unit continuously locks the synchronization clock (e.g., the time slot reference of WBS broadband voice 16kHz) sent by the communication terminal (mobile phone) as the Target Clock.

[0055] 2. Detect the micro slope (high and low watermarks of the FIFO) of the first and second uplink data streams when filling the local buffer.

[0056] 3. When the target source to be reverse-switched (such as an external large-diaphragm microphone, configured to 48kHz but actually operating at 47.999kHz due to crystal oscillator error) is detected to be inconsistent with the Target Clock, ASRC will activate a polyphase filter resampler.

[0057] 4. By using hardware adaptive interpolation (for slower clocks) or smooth data point extraction (for faster clocks), the length of the target source data packets is forced to be a microframe that is exactly the same as the standard step size required by the current link.

[0058] This hardware-level data alignment ensures that the system's dual-source dynamic injection can maintain a perfect, microsecond-level match at the data seams regardless of the type of source device. Example 5: Absolute physical dimensionality reduction isolation for low-level multi-track concurrent extraction and storage

[0059] Based on the uninterrupted hot switching and clock compensation established in the above embodiments, the present invention further proposes a system capable of outputting 100% absolutely physically isolated dual-track audio as a base for subsequent artificial intelligence analysis.

[0060] Multi-channel audio in existing communication terminals (such as meeting recording apps) is often generated by extracting the audio stream after the mixer, and its left and right tracks are not protected at the physical source.

[0061] In this system, the underlying layer of the concurrent network stack processing module is equipped with a split copy pipeline (such as a direct branch point of the I2S data line).

[0062] On the one hand, before the downlink voice baseband packets pass through the concurrent network stack processing module inside the main control device and are sent to the digital-to-analog converter (DAC), the split copy pipeline intercepts them as the first clean data stream containing only the other party's voice (such as pre-allocated to the left channel).

[0063] On the other hand, when the baseband packets from the local acquisition device (such as an external lavalier microphone) are about to be packaged and sent to the communication terminal inside the main control hardware, and before the underlying encryption and compression are performed, the split copy pipeline also intercepts them and uses them as a second clean data stream containing only its own voice (such as pre-allocated to the right channel).

[0064] Because the two channels are extracted completely independently on serial or parallel physical buses, and their respective write clocks are synchronized by the ASRC control in Example 4, their physical isolation reaches 100%. The audio streams of these two tracks are synchronously merged into the local non-volatile storage module, and the final audio file naturally contains acoustic metadata tags bound to physical roles. When subsequently connecting to large-scale cloud-based language AI, there is no need for the extremely time-consuming Blind Source Separation, resulting in a significant improvement in the transcription success rate of overlapping argument segments in complex scenarios. Example 6: Detailed Explanation of Network Configuration and Dynamic Injection Points for External Microphones

[0065] The specific combination and connection logic of the second uplink voice source (i.e., the externally accessed pickup peripheral) as specifically referred to above and in this system will now be explained in further detail.

[0066] In actual product applications, depending on the needs of different professional scenarios, the main control device may need to be connected to different types of pickup devices with different interface forms, such as directional dynamic microphones, wired lavalier electret microphones, or another wireless microphone unit with high-specification noise reduction capabilities.

[0067] At the physical access subsystem level, the main control device also includes a multimodal physical interface module in its circuit structure.

[0068] The multimodal physical interface module includes at least the following:

[0069] A) Analog audio transceiver channel with a separate analog-to-digital converter (ADC) front end and phantom power control pin for direct connection to a standard 3.5mm or dedicated multi-pin wired analog microphone.

[0070] B) A digital audio bus interface (such as a USB / Type-C Audio Class controller or a dedicated I2S expansion interface) for bridging high-definition digital microphone arrays with independent digital signal processors.

[0071] When an external device (e.g., an external wired microphone) is physically connected to the main control device.

[0072] Phase 1: Initialization and Wake-up. The peripheral detection unit within the main control device detects level changes or protocol enumeration events, activates the audio front-end (such as a specific ADC register segment or USB endpoint) where the external microphone is located, and begins receiving the raw audio data pool. At this time, this data is not sent to the mobile phone, but only exists in the local second buffer address (such as the local acoustic codec buffer mentioned in step 304 of Example 3).

[0073] Phase Two: Button Binding and Logic Registration. The logic pins (GPIO or interrupt pins) of the physical PTT (Push-To-Talk) button or external wired control command module on the main control device are mapped to the uplink data decision module described in Example 2.

[0074] Phase 3: Transient Dial-Up Injection. Within a very short interval when the user presses the PTT button, the system determines the trigger is valid. Then, following step 304 of Example 3, the system uses DMA or a bus routing controller to precisely shift the uplink baseband source read pointer towards the communication terminal from the first buffer (headphone audio) which is in a suspended waiting state to the aforementioned second buffer, which has been continuously filled by the external microphone. Thus, the high-definition digital audio sequence captured by the external microphone is seamlessly integrated into the Bluetooth HFP / USB uplink channel that is maintaining contact with the mobile phone, completing the closed loop of external audio source takeover of the core communication link. Example 7: Advanced Always-On Switching Mechanism Based on VAD and Voice Wake-up

[0075] In Example 6, the condition for triggering the source pointer switching is a physical button press (PTT). However, in certain high-frequency interaction scenarios, users may need to completely free their hands. Therefore, this example further proposes a normally open injection mode based on environmental acoustic analysis.

[0076] In this implementation, the uplink data decision module of the main control device is not only controlled by the physical GPIO level transition, but also deeply integrated with the built-in or external low-power voice activity detection (VAD) chip unit or wake-up word engine.

[0077] Front-end resident detection network: The local acquisition unit (external second microphone array) continues to operate and continuously send acoustic feature sequences to its associated ultra-low power buffer after the main control device is powered on, even if the system is in a sleep-keep state. This feature sequence is directly sent to the VAD detection layer.

[0078] Hot detection: When the user starts to speak or utters a voice (or a specific wake word) above a certain threshold loudness, the VAD detection layer instantly outputs a soft interrupt signal to the uplink data decision module.

[0079] This soft interrupt is equivalent to the physical interrupt operation of pressing the PTT in Embodiment 6, directly driving the aforementioned instantaneous atomic operation switching of the DMA pointer (see step 304 in Embodiment 3); and after a settable delay window (such as a 500ms trail retention time) after the VAD indicates the end of the voice, the system automatically switches the DMA read pointer back to the original first uplink source (communication headset) spontaneously by hardware.

[0080] This pure voiceprint / threshold driving mechanism, which requires no physical button intervention, enables the main control device to achieve the same millisecond-level communication link resource preemption and hot-switching effect as manual PTT while presenting an always-on, imperceptible state. This greatly improves the human-computer interaction efficiency of smart wearable solutions.

Claims

1. A communication system based on a mediator dual-proxy architecture, characterized in that, include: The system comprises a first communication node serving as the terminal endpoint, a second communication node serving as both the sound source and the basic pickup receiver, and a master control device logically connected in series between the first and second communication nodes. The master control device is equipped with a baseband concurrent processing module for simultaneously processing and transmitting digital audio messages while maintaining a first physical connection between the master control device and the first communication node, and a second physical connection between the master control device and the second communication node. The master control device has an external interface independent of the first and second physical connections, configured to acquire local input as a third acoustic source. The master control device includes an uplink data decision module and a direct memory access (DMI) or equivalent internal data bus. The internal data bus is configured with a first buffer address and a backup buffer address for assembling an uplink audio message. The first buffer address carries an audio feature stream from the second communication node, and the backup buffer address carries an audio feature stream from the third acoustic source. The uplink data decision module is configured, under autonomous control, to respond to a local trigger event and, within a small critical gap, perform an irreversible atomic operation to forcibly switch the data read pointer of the uplink audio message from the first buffer address to the backup buffer address or vice versa. The underlying firmware instruction set of the master control device maintains system-level concealment at the link layer to the first communication node during the switching operation, exhibiting the following continuity characteristics: (a) The synchronous connection-oriented (SCO / eSCO) data link carrying the uplink audio message maintains an absolutely continuous sequence number without any physical channel address shift or active disconnection and reconnection of the link; (b) Ensure that no upper-layer application protocol handshake signaling (including AT+BIA instructions) that requires changing the current audio communication endpoint is sent to the first communication node, thereby avoiding triggering the reassignment process of the audio routing policy (Audio Policy) underlying manager of the first communication node's operating system.

2. The communication system according to claim 1, characterized in that, The external interface includes a wired contact pin circuit interface or a wireless signal bridging module that operates independently in a dedicated short-range radio frequency band to acquire the local input. The peripheral module corresponding to the third acoustic source acquired through the external interface, located in an independent physical time domain, is independently driven by a hardware clock and generates the local input.

3. The communication system according to claim 1, characterized in that, The baseband concurrent processing module of the main control device includes an asynchronous sampling rate compensation mechanism (ASRC) layer. The asynchronous sampling rate compensation mechanism layer is hard-wired to perform bit-depth adaptive hardware resampling on the data sequence originating from the third acoustic source. This process is limited by maintaining the bit depth of the new payload after the switch strictly matching the fixed payload length of the synchronous connection-oriented (SCO / eSCO) microframe. This serves as a necessary physical prerequisite to prevent the reference clock from crashing and overflowing when the first buffer address is switched to the backup buffer address.

4. The communication system according to claim 1, characterized in that, The first communication node is an initiating or receiving host smart device with cellular or LAN voice communication capabilities; the second communication node is a wireless or wired headset device. Correspondingly, the baseband concurrent processing module of the main control device is configured to simulate hands-free (HF) device specifications to establish the first physical connection, and to isolate simulated audio gateway (AG) device specifications to establish the second physical connection.

5. The communication system according to claim 1, characterized in that, The baseband concurrent processing module of the main control device is also equipped with a bypass pipeline. The bypass pipeline is configured to intercept and copy the audio baseband packets in the downlink direction as a first physical data stream before they are sent to the digital-to-analog converter; A copy is intercepted as a second physical data stream when the audio baseband packets in the uplink direction are assembled but before physical layer transmission encryption is implemented. The first physical data stream and the second physical data stream do not perform digital mixing operations. They are aligned by hardware time synchronization stamps and written in parallel to an independent non-volatile memory unit that is not directly accessible via network addressing and is embedded in the motherboard inside the main control device.

6. The communication system according to claim 1, characterized in that, The locally triggered event manifests as an interrupt request generated by a physical state transition of the interactive sensing unit composed of the main control device and its associated components; the physical state transition specifically includes any single-dimensional response mode or a logical combination thereof: (A) Touch response mode: The abrupt change is generated by the capacitive signal triggered by the touch or slide on the system housing and the change in the touch torque. (B) Physical motion posture mode: Over-limit vector generated by the built-in inertial measurement unit (IMU) when the impact, lifting or displacement acceleration exceeds the static threshold; (C) External magnetic / photoelectric proximity mode: Level flipping when a device with Hall effect sensing function or an infrared sensor detects that the object is detached from the base magnetic attraction or approaches an obstacle; (D) Dedicated out-of-band channel remote control mode: Virtual control level commands received by an independent radio frequency bypass (such as a low-power Bluetooth channel) that does not carry the primary communication load of the first physical connection, and thrown back from the auxiliary smart screen or the matching derivative button accessories. (E) Physical contact loop mode: a pin transition hardware-level interrupt generated by the pressing action of a microswitch or contact on the dedicated communication line or wire control module. Alternatively, a soft interrupt acoustic vector is thrown by a sound activity detection (VAD) engine and wake word recognition accelerator that are hard-coded and run by the main control device when the captured energy waveform exceeds a threshold.

7. The communication system according to claim 1, characterized in that, The autonomous control state further includes a shielding and isolation configuration to prevent remote data acquisition. When the master control device intercepts downlink signaling packets from the first communication node attempting to suspend, interrupt, or mute remote data acquisition behavior via the communication link, the interception manager directly cuts off the downlink transmission and throws a fatal deceptive positive response packet back to the upper layer through a simulated transmission loop to complete a fake handshake, forcing the underlying baseband to continue maintaining the current operation of all data reading buffer state machines without being affected by external hang-up commands.

8. A mid-level power agent hardware master control device, characterized in that, It is applied to the cross-acoustic environment system as defined in any one of claims 1 to 7, and includes a printed circuit board assembly with a dedicated micro-instruction assembly package and a memory routing matrix set. When powered on, it passively parses the downlink control tree and autonomously and actively implements the aforementioned pointer overlay and continuous frame suppression strategy to form an intercepting physical node.

9. A processing method for achieving seamless audio source switching without inducing topology reconnection, characterized in that, Deployed and operating within an independent master control hardware device, the specific timing includes: Initializing and taking over two unrelated basic short-range communication connection processes for the call host and the voice unit. While maintaining the existing downlink communication receive address without interruption, new physical acoustic electrical parameters are intercepted on a parallel, undisclosed interface and injected into a new, independent spare memory segment. The system's interrupt trigger pin is polled with the highest priority clock. After capturing the interrupt pulse and forcibly stopping the previous microsecond loop, the pointer for the payload of the next uplink data packet to be transmitted is extracted using a memory address accessor and offset towards the new spare memory segment. Within the time frame and packet transmission cycle of this replacement completion, any application layer (AT command group) attribute events indicating that the microphone role topology has been replaced are blocked or suppressed from being pushed to or confirmed by the call host, and the underlying connection packet sequence number (SCO / eSCO Sequence Number) is kept strictly continuous.

10. A chip-level non-volatile medium with a driver binary file engraved thereon, wherein the arrangement of operating logic potentials stored thereon, when acquired and decompressed into a running state by an upstream computing unit, necessarily forces the master device to implement a method as defined in claim 9, including all steps and constraints.

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