A central control method and system for synchronously playing audio in multiple rooms of a hotel

Abstract

The present invention provides a central control method and system for synchronously playing audio in multiple rooms of a hotel, which is applied to the field of data synchronization processing; the present invention constructs an audio stream transmission path diagram based on a network topology preset by an audio routing table, and the system can intuitively view the audio stream transmission route of each room, identify and optimize the links that may cause delays, and can not only achieve synchronous playback of audio across rooms, but also ensure that audio signals are transmitted with minimum delay, thereby improving audio quality. At the same time, by dynamically adjusting the compression ratio of audio encoding, the system can reduce network bandwidth pressure while maintaining audio quality, thereby avoiding audio transmission interruption or quality degradation caused by insufficient network bandwidth.

Description

Technical Field

[0001] The present invention relates to the field of data synchronization processing, and in particular to a central control method and system for synchronously playing audio in multiple rooms of a hotel. Background Art

[0002] In hotel environments, there is an increasing need for synchronized audio playback in multiple rooms to enhance the guest experience, support conference activities or provide background music.

[0003] However, the use of hotel rooms often changes. For example, small meeting rooms are merged into large conference halls or temporarily divided into independent spaces. Traditional statically configured audio systems find it difficult to flexibly adjust the network configuration and playback logic of audio equipment according to actual needs. Summary of the invention

[0004] The present invention aims to solve the problem that it is difficult for traditional statically configured audio equipment to flexibly adjust the network configuration and playback logic of audio equipment according to actual needs, and provides a central control method and system for synchronously playing audio equipment in multiple rooms of a hotel.

[0005] The present invention adopts the following technical means to solve the technical problem:

[0006] The present invention provides a central control method for synchronously playing audio in multiple rooms of a hotel, comprising:

[0007] Based on the to-be-synchronized area preset by the hotel terminal, identifying the corresponding room synchronization configuration from the to-be-synchronized area;

[0008] Determining whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal;

[0009] If not, then according to the preset room layout information of the hotel terminal, apply the preset network scanning protocol to obtain the real-time audio stream of the room synchronization configuration, dynamically merge the audio stream path of the room synchronization configuration according to the real-time audio stream, identify the audio requirements preset in the room synchronization configuration, and adaptively adjust the audio routing table of the hotel terminal according to the audio requirements;

[0010] determining whether the audio routing table detects an audio delay;

[0011] If detected, based on the network topology preset in the audio routing table, an audio stream transmission path diagram of the room synchronization configuration is constructed, and according to the synchronization signal of the audio stream transmission path diagram, the playback delay of the room synchronization configuration is collected, and the preset audio test tool is applied to obtain the error data of the playback delay, and according to the error data, the compression ratio of the audio encoding is dynamically reduced.

[0012] Furthermore, before the step of applying a preset network scanning protocol to obtain the real-time audio stream of the room synchronization configuration, the step further includes:

[0013] Based on the audio source device synchronously configured in the room, obtaining the audio buffer preset by the hotel terminal;

[0014] Determining whether the audio buffer is consistent with the current network status of the hotel terminal;

[0015] If not, identify the change in network quality of the hotel terminal, dynamically adjust the audio cache of the audio buffer according to the change in network quality, and adaptively adjust the audio output content of the hotel terminal according to the environmental conditions preset in the room synchronization configuration, wherein the network quality change specifically includes packet loss, delay and jitter, the environmental conditions specifically include noise level, population density and room size, and the audio output content specifically includes audio source content, audio output quality and ambient sound perception.

[0016] Further, the step of identifying the preset audio requirements of the room synchronization configuration and adaptively adjusting the audio routing table of the hotel terminal according to the audio requirements also includes:

[0017] Based on the audio type preset by the user through the hotel terminal, identifying the audio format of the room synchronization configuration, wherein the audio type specifically includes background music, voice broadcast and entertainment content;

[0018] Determining whether the audio format matches the audio requirement;

[0019] If not, the room synchronization configuration is divided into various audio playback areas according to the room layout information, and the playback strategy of the audio routing table is coordinated according to the audio playback area, wherein the playback strategy specifically includes single loop, random playback and timed playback.

[0020] Furthermore, the step of constructing the audio stream transmission path diagram of the room synchronization configuration based on the network topology preset in the audio routing table further includes:

[0021] Based on the device nodes involved in the audio stream transmission, the connection relationship paths of the device nodes are collected, wherein the device nodes specifically include audio equipment, routers and switches, and the connection relationship paths specifically include directly connected devices, network segment distribution and bandwidth information;

[0022] Determine whether the connection relationship path can meet the bandwidth requirement;

[0023] If possible, the device nodes are layered according to a preset network structure hierarchy, and the functional classification labels of the device nodes are divided. Based on the functional classification labels, the communication paths between devices are detected, wherein the network structure hierarchy specifically includes a core layer, an access layer, and an edge layer, and the functional classification labels specifically include an audio output node, a network forwarding node, and a data forwarding node.

[0024] Furthermore, the step of determining whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal further includes:

[0025] Based on the historical synchronization configuration, detecting the room layout information provided by the hotel terminal to the user;

[0026] Determining whether the room layout information is the same as the historical synchronization configuration;

[0027] If not, the power supply location in the hotel room is identified based on the historical synchronization configuration, and the location layout of the sound source device in the hotel room is dynamically adjusted based on the power supply location. The synchronization delay of the sound source device is recalibrated through the distance change of the location layout.

[0028] Further, the step of determining whether the audio routing table detects audio delay further includes:

[0029] Based on the transmission process of the audio signal from the audio source device to the target device, marking the audio packet timestamps of the audio signal between various nodes;

[0030] Determining whether the sending time and the receiving time of the audio packet timestamp are within the same preset interval;

[0031] If not, then the audio delay of the audio packet timestamp is calculated according to the sending time and the receiving time, the tolerant delay value from the audio source device to the target device is monitored, and based on the tolerant delay value, the delay compensation measures preset by the hotel terminal are triggered, wherein the delay compensation measures specifically include adjusting the routing path of the audio stream, dynamically allocating the bandwidth of the audio stream, and adjusting the buffer of the audio stream.

[0032] Furthermore, the step of identifying the corresponding room synchronization configuration from the area to be synchronized based on the area to be synchronized preset by the hotel terminal further includes:

[0033] Based on the audio synchronization area pre-selected by the user through the hotel terminal, identifying at least two corresponding hotel rooms from the audio synchronization area;

[0034] Determining whether the hotel guest room can detect a preset audio linkage;

[0035] If not, check the audio linkage configuration between the hotel rooms, identify the abnormal type of the audio linkage according to the audio linkage configuration, and review the linkage availability of the hotel rooms through the hotel terminal based on the abnormal type, wherein the abnormal type specifically includes device connection abnormality, audio signal transmission abnormality, and network and hardware abnormality.

[0036] The present invention also provides a central control system for synchronously playing audio in multiple rooms of a hotel, comprising:

[0037] An identification module, used for identifying a corresponding room synchronization configuration from the area to be synchronized based on the area to be synchronized preset by the hotel terminal;

[0038] A judgment module, used to judge whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal;

[0039] An execution module, for, if not, applying a preset network scanning protocol according to the room layout information preset by the hotel terminal, obtaining the real-time audio stream of the room synchronization configuration, dynamically merging the audio stream path of the room synchronization configuration according to the real-time audio stream, identifying the audio requirement preset by the room synchronization configuration, and adaptively adjusting the audio routing table of the hotel terminal according to the audio requirement;

[0040] A second determination module is used to determine whether the audio routing table detects audio delay;

[0041] The second execution module is used to, if detected, construct an audio stream transmission path diagram of the room synchronization configuration based on the network topology preset in the audio routing table, collect the playback delay of the room synchronization configuration according to the synchronization signal of the audio stream transmission path diagram, apply a preset audio test tool to obtain the error data of the playback delay, and dynamically reduce the compression ratio of the audio encoding according to the error data.

[0042] Furthermore, it also includes:

[0043] An acquisition module, used for acquiring an audio buffer preset by the hotel terminal based on an audio source device synchronously configured in the room;

[0044] A third judgment module is used to judge whether the audio buffer area conforms to the current network status of the hotel terminal;

[0045] The third execution module is used to identify the change of network quality of the hotel terminal if not, dynamically adjust the audio cache of the audio buffer according to the change of network quality, and adaptively adjust the audio output content of the hotel terminal according to the preset environmental conditions of the room synchronization configuration, wherein the network quality change specifically includes packet loss, delay and jitter, the environmental conditions specifically include noise level, population density and room size, and the audio output content specifically includes audio source content, audio output quality and ambient sound perception.

[0046] Furthermore, the execution module also includes:

[0047] An identification unit, configured to identify the audio format of the room synchronization configuration based on the audio type preset by the user through the hotel terminal, wherein the audio type specifically includes background music, voice broadcast and entertainment content;

[0048] A judging unit, configured to judge whether the audio format matches the audio requirement;

[0049] The execution unit is used to divide the room synchronization configuration into various audio playback areas according to the room layout information, and coordinate the playback strategy of the audio routing table according to the audio playback area, wherein the playback strategy specifically includes single loop, random playback and timed playback.

[0050] The present invention provides a central control method and system for synchronously playing audio in multiple rooms of a hotel, which has the following beneficial effects:

[0051] The present invention constructs an audio stream transmission path diagram based on the network topology preset in the audio routing table. The system can intuitively view the audio stream transmission routes of each room, identify and optimize the links that may cause delays, and can not only achieve synchronous playback of audio across rooms, but also ensure that the audio signal is transmitted with the lowest delay, thereby improving the audio quality. At the same time, by dynamically adjusting the compression ratio of the audio encoding, the system can reduce the pressure on the network bandwidth while maintaining the audio quality, thereby avoiding audio transmission interruption or quality degradation caused by insufficient network bandwidth. BRIEF DESCRIPTION OF THE DRAWINGS

[0052] Figure 1 A flow chart of an embodiment of a central control method for synchronously playing audio in multiple rooms of a hotel according to the present invention;

[0053] Figure 2 The present invention is a structural block diagram of an embodiment of a central control system for synchronously playing audio in multiple rooms of a hotel. DETAILED DESCRIPTION

[0054] It should be understood that the specific embodiments described herein are only used to explain the present invention and are not intended to limit the present invention. The implementation of the objectives, functional features and advantages of the present invention will be further described in conjunction with the embodiments and with reference to the accompanying drawings.

[0055] The following will be combined with the drawings in the embodiments of the present invention to clearly and completely describe the technical solutions in the embodiments of the present invention. Obviously, the described embodiments are only part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the present invention.

[0056] Reference Figure 1 , is a central control method for synchronously playing audio in multiple rooms of a hotel in one embodiment of the present invention, comprising:

[0057] S1: Based on the to-be-synchronized area preset by the hotel terminal, identifying the corresponding room synchronization configuration from the to-be-synchronized area;

[0058] S2: Determine whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal;

[0059] S3: If not, then according to the preset room layout information of the hotel terminal, apply the preset network scanning protocol to obtain the real-time audio stream of the room synchronization configuration, dynamically merge the audio stream path of the room synchronization configuration according to the real-time audio stream, identify the audio requirements preset in the room synchronization configuration, and adaptively adjust the audio routing table of the hotel terminal according to the audio requirements;

[0060] S4: Determine whether the audio routing table detects audio delay;

[0061] S5: If detected, based on the network topology preset in the audio routing table, construct an audio stream transmission path diagram of the room synchronization configuration, collect the playback delay of the room synchronization configuration according to the synchronization signal of the audio stream transmission path diagram, apply a preset audio test tool to obtain the error data of the playback delay, and dynamically reduce the compression ratio of the audio encoding based on the error data.

[0062] In this embodiment, the system identifies the corresponding room synchronization configuration from the audio synchronization areas pre-set by the hotel terminal, and then determines whether the room synchronization configuration set by the user matches the historical synchronization configuration pre-recorded by the hotel terminal to execute the corresponding steps; for example, when the system determines that the room synchronization configuration set by the user can match the historical synchronization configuration pre-recorded by the hotel terminal, the system will consider that the audio linkage requirement set by the user is consistent with the existing configuration of the hotel terminal, and the system can directly execute the operation corresponding to the historical configuration to avoid unnecessary reconfiguration or adjustment, thereby improving efficiency and reducing potential errors. The system will confirm that the room synchronization configuration set by the user is completely consistent with the audio stream, device connection, routing table and other information of the historical configuration to ensure that there are no differences, and ensure that the two match by comparing and checking key information such as device ID, audio routing, audio source, room layout, etc. At the same time, once the match is confirmed, the system can directly load the historical synchronization configuration pre-recorded by the hotel terminal, which means that there is no need to reconfigure or adjust. Generate audio stream paths, audio routing tables and other contents, and directly apply the parameters in the historical configuration to reduce the time for calculation and adjustment. Before executing the historical configuration, the system can quickly scan the hotel's network environment to check whether network indicators such as bandwidth and latency meet the requirements of the historical configuration. If the network conditions do not match, the system can optimize the network environment through intelligent adjustment or prompt the user to ensure the smoothness of audio synchronization. For example, when the system determines that the room synchronization configuration set by the user cannot match the historical synchronization configuration pre-recorded by the hotel terminal, the system will consider that the audio linkage requirements set by the user are inconsistent with the existing configuration of the hotel terminal, and the system cannot directly perform operations that do not correspond to the historical configuration. The system will apply the pre-set network scanning protocol according to the room layout information pre-set by the hotel terminal to obtain the real-time audio stream of the room synchronization configuration, and dynamically merge the audio stream path of the room synchronization configuration based on these real-time audio streams, identify the audio requirements of the room synchronization configuration, and adaptively adjust the audio routing table of the hotel terminal based on the audio requirements.By applying a pre-set network scanning protocol and obtaining real-time audio streams, the system can dynamically identify and adapt to the audio needs of the room. Whether it is increasing or decreasing the number of rooms, adjusting the volume or changing the audio source, the system can make adjustments based on real-time data. Even if the room synchronization configuration set by the user does not match the historical configuration, the system can flexibly respond to different configuration requirements by obtaining audio streams in real time and dynamically merging audio paths, avoiding the limitations of traditional static configurations that cannot be adapted. At the same time, through the analysis of real-time audio streams, the system can identify the audio needs of each room and intelligently merge audio stream paths, which not only improves the transmission efficiency of the audio stream, but also avoids the problems of audio stream conflicts or unreasonable paths that may occur in traditional configurations. In addition, by dynamically merging audio stream paths and optimizing audio routing tables, the system can effectively ensure audio quality and playback synchronization, avoiding problems such as sound quality degradation and excessive delay that may occur in traditional systems. The system then determines whether the audio routing table detects audio delay to execute the corresponding steps. For example, when the system determines that the audio routing table to the hotel terminal does not detect audio delay, the system will consider that the current audio stream transmission is smooth and the audio synchronization effect of each room in the sound system meets the expected During the period, there is no audio delay caused by network transmission problems or improper device configuration. The system will record the current audio routing table configuration and the transmission status of the audio stream to provide reference data for subsequent room synchronization configuration. If the user makes new configurations or modifications in the future, the system can quickly load the historical configuration to ensure that the new configuration still meets the synchronization requirements. At the same time, since there is no audio delay problem, the system does not need to adjust the compression ratio of the audio encoding, which can avoid affecting the sound quality due to excessive or insufficient compression. In order to ensure the long-term stability of the system, the system should continue to monitor the transmission quality of the audio stream and detect in real time whether there is a new delay or audio quality degradation. Through continuous monitoring, the system can respond to any possible problems in a timely manner; for example, when the system determines that the audio routing table of the hotel terminal can detect audio delay, the system will think that the current audio stream transmission is not smooth. The system will build an audio stream transmission path diagram for the room synchronization configuration based on the network topology pre-set by the audio routing table, collect the playback delay of the room synchronization configuration according to the synchronization signal of the audio stream transmission path diagram, apply the pre-set audio test tool, obtain the error data of the playback delay, and dynamically reduce the compression ratio of the audio encoding based on these error data;By constructing an audio stream transmission path diagram and optimizing the path of the audio stream according to the diagram, the system can find out the network bottleneck or device blockage that causes audio delay. After dynamically adjusting the audio stream path, it can reduce conflicts and blockages during transmission and make the audio signal smoother. At the same time, through synchronous signal acquisition and playback delay monitoring, the system can accurately understand the audio playback delay in each room, and judge whether there is a link that needs to be optimized through data feedback. The collected real-time data can help the system make adjustments at critical moments to improve the quality of the audio stream. In addition, according to the error data, the system can dynamically reduce the compression ratio of audio encoding. By reducing the compression ratio, the system can reduce the distortion that may be introduced during the compression process, especially the sound quality loss under high compression conditions, so as to ensure the quality and clarity of the audio, especially in poor network environments or limited bandwidth. ;

[0063] In this embodiment, before the step of applying a preset network scanning protocol to obtain the real-time audio stream of the room synchronization configuration, the step further includes:

[0064] S301: Based on the audio source device synchronously configured in the room, obtain the audio buffer preset by the hotel terminal;

[0065] S302: Determine whether the audio buffer area meets the current network status of the hotel terminal;

[0066] S303: If not, identifying the change in network quality of the hotel terminal, dynamically adjusting the audio cache of the audio buffer according to the change in network quality, and adaptively adjusting the audio output content of the hotel terminal according to the environmental conditions preset in the room synchronization configuration, wherein the network quality change specifically includes packet loss, delay and jitter, the environmental conditions specifically include noise level, population density and room size, and the audio output content specifically includes audio source content, audio output quality and ambient sound perception.

[0067] In this embodiment, the system obtains the audio buffer pre-set in the hotel terminal based on the audio source device synchronously configured in the room, and then the system determines whether the audio buffer meets the current network status of the hotel terminal to execute the corresponding steps; for example, when the system determines that the audio buffer pre-set in the hotel terminal can meet the current network status of the hotel terminal, the system will believe that the network can smoothly carry the transmission of the current audio stream, and the capacity and performance of the audio buffer are sufficient to cope with the stable and smooth transmission of audio data. The system will continue to use the current audio routing configuration, network bandwidth and buffer size without adjusting the audio stream path, compression rate or other network parameters, which reduces system resource consumption and ensures the high efficiency of audio playback. At the same time, the adaptability of the buffer ensures the smooth transmission of audio, and no additional adjustment or processing is required to solve the problems caused by the network. The audio effect can be consistent and maintain high quality. And if the audio buffer matches the network status, the system can confirm that the transmission path of the audio stream and the room audio linkage mechanism have stabilized, and the audio data is synchronously transmitted between the rooms without delay. The system can continue to play the audio content according to the preset room layout without the need for other complex manual adjustments; for example, when the system determines that the audio buffer pre-set in the hotel terminal cannot meet the current network status of the hotel terminal, the system will consider that the network cannot carry the transmission of the current audio stream. The system will identify the changes in the network quality of the hotel terminal, which specifically include packet loss, delay and jitter. According to the changes in the network quality, the audio cache of the audio buffer is dynamically adjusted, and the audio output content of the hotel terminal is adaptively adjusted according to the environmental conditions pre-set in the room synchronization configuration, which specifically include noise level, population density and room size. The audio output content specifically includes audio source content, audio output quality and ambient sound perception;The system can optimize the audio buffer in a targeted manner through real-time detection of packet loss, delay and jitter, thereby reducing audio interruption, freeze and loss caused by network problems. This real-time network optimization can ensure the complete transmission and seamless playback of audio content. At the same time, it can dynamically adjust the audio buffer according to the real-time network quality, and can also adaptively adjust the audio output content in combination with the environmental conditions of the room. This highly intelligent adjustment method can ensure that even if the network conditions or environmental conditions change, the sound effect is always optimal. By dynamically adjusting the audio settings according to the real-time network conditions and room environment changes, the system can avoid overusing audio resources or unnecessary waste of resources. For example, when the network conditions are good, the system can reduce the size of the buffer and reduce the complexity of audio encoding. When the network is unstable, the system can increase the buffer and reduce the compression ratio to improve the stability of the audio stream. This dynamic adjustment mechanism reduces unnecessary operations and resource consumption. Especially when the network quality is unstable, the system can automatically optimize the transmission path and buffer mechanism of the audio stream to avoid excessive resource occupation or overload of processing capacity. ;

[0068] In this embodiment, the step S3 of identifying the audio requirement preset in the room synchronization configuration and adaptively adjusting the audio routing table of the hotel terminal according to the audio requirement further includes:

[0069] S31: Based on the audio type preset by the user through the hotel terminal, identifying the audio format of the room synchronization configuration, wherein the audio type specifically includes background music, voice broadcast and entertainment content;

[0070] S32: Determine whether the audio format matches the audio requirement;

[0071] S33: If not, the room synchronization configuration is divided into various audio playback areas according to the room layout information, and the playback strategy of the audio routing table is coordinated according to the audio playback area, wherein the playback strategy specifically includes single loop, random playback and timed playback.

[0072] In this embodiment, the system identifies the audio format of the room synchronization configuration based on the audio type pre-set by the user through the hotel terminal, the audio type specifically includes background music, voice broadcast and entertainment content, and then the system determines whether these audio formats match the audio type preset in the hotel terminal to execute the corresponding steps; for example, when the system determines that the audio format can match the audio type preset in the hotel terminal, the system will consider that the identified audio stream format is consistent with the audio requirement preset by the system and meets the usage scenario of the current room and the requirements of the user. The system will route the audio signal to each audio output device (such as a speaker or a TV, etc.) according to the current room synchronization configuration. For example, for background music, the system can ensure that the audio signal is synchronized by area Distribution, adapt to the playback needs of different rooms or areas. For voice broadcasts, the audio stream can be transmitted to the room that needs to be broadcast first, and the clarity and delay are minimized. For entertainment content, the system can adjust the loudness and effect of the audio output according to the room size and equipment configuration to provide an immersive experience, and monitor the transmission status of the audio stream in the network in real time to ensure that there are no delays, packet loss or distortion. If network delays or quality problems are detected, the system can automatically perform buffering, compression ratio adjustment or audio stream redirection to ensure audio stability and quality; for example, when the system determines that the audio format of the room synchronization configuration cannot match the audio type preset on the hotel terminal, the system will consider the recognized audio stream format and If the preset audio requirements are inconsistent, the system will divide the room synchronization configuration into various audio playback areas according to the room layout information, and coordinate the playback strategy of the audio routing table according to different audio playback areas. The playback strategy specifically includes single song loop, random playback and timed playback; the system can set different playback strategies for different areas by dividing the room synchronization configuration into multiple audio playback areas. This regionalized control method allows the system to achieve flexible audio playback and management in different rooms or areas even if the audio format cannot completely match the preset audio type. At the same time, by adjusting the playback strategy and regional division, the system can avoid playback conflicts or instability caused by inconsistent audio formats. Different rooms use Different playback strategies can reduce the problems caused by audio stream asynchrony or incompatible audio formats. Through flexible playback strategies, the system can automatically adjust the playback mode of audio content according to the needs of different areas of the hotel to ensure that the audio content is highly matched with the scene. For example, during meal times, the audio system in the restaurant area can play light music or background music, while the conference room may play news or broadcasts. Such intelligent playback control greatly improves the user experience. By dynamically adjusting audio playback according to playback strategies and regional needs, the system can automatically shut down audio equipment or switch to low power consumption mode when needed to avoid energy waste. For example, during non-peak hours, audio equipment can choose timed playback mode to automatically stop playback and reduce energy consumption.

[0073] In this embodiment, the step S5 of constructing the audio stream transmission path diagram of the room synchronization configuration based on the network topology preset in the audio routing table further includes:

[0074] S51: Based on the device nodes involved in the audio stream transmission, collecting the connection relationship paths of the device nodes, wherein the device nodes specifically include audio equipment, routers and switches, and the connection relationship paths specifically include directly connected devices, network segment distribution and bandwidth information;

[0075] S52: Determine whether the connection relationship path can meet the bandwidth requirement;

[0076] S53: If possible, the device nodes are layered according to a preset network structure hierarchy, and functional classification labels of the device nodes are divided. Based on the functional classification labels, the communication paths between devices are detected, wherein the network structure hierarchy specifically includes a core layer, an access layer, and an edge layer, and the functional classification labels specifically include an audio output node, a network forwarding node, and a data forwarding node.

[0077] In this embodiment, the system is based on the device nodes involved in the audio stream transmission, the device nodes specifically include audio equipment, routers and switches, and collects the connection relationship paths of these device nodes, the connection relationship paths specifically include directly connected devices, network segment distribution and bandwidth information, and then the system determines whether these connection relationship paths can meet the bandwidth requirements to execute the corresponding steps; for example, when the system determines that the connection relationship path to the device node cannot meet the bandwidth requirements, the system will believe that the current network resources are insufficient to support high-quality audio stream transmission, which may cause problems such as audio stream freezes, delays or packet loss, thereby affecting the normal operation of the audio equipment. The system will identify the location of the current bandwidth bottleneck by analyzing the connection relationship between the device nodes. The bottleneck nodes are usually devices with insufficient network transmission capabilities (such as routers, switches or certain audio equipment), and these nodes need to be optimized. At the same time, the bandwidth allocation is dynamically adjusted according to priority and usage requirements. For example, priority is given to ensuring the transmission bandwidth of audio streams, limiting the bandwidth usage of non-critical traffic (such as controlling the bandwidth of web browsing or video traffic), and if the network bandwidth is limited, the system can appropriately adjust the compression ratio of the audio stream to reduce bandwidth usage, such as using lower audio quality settings or adjusting the encoding method to reduce the bandwidth required for transmission; for example, when the system determines that the connection path to the device node can meet the bandwidth requirements, the system will then believe that the current network resources are sufficient to support high-quality audio stream transmission, and the system will classify these device nodes according to the pre-set network structure hierarchy, which specifically includes the core layer, access layer, and edge layer, and divide the device nodes into functional categories and labels. The functional classification labels are specific Including audio output nodes, network forwarding nodes and data forwarding nodes, according to different functional classification labels, the communication path between devices is detected; the system can clearly define each part of the network by layering the devices according to the core layer, access layer and edge layer, and formulate corresponding bandwidth requirements, delay restrictions and fault tolerance mechanisms for each layer. This hierarchical structure helps to improve the management efficiency of the network and ensure the coordination and cooperation between the layers. At the same time, by marking the device nodes as audio output nodes, network forwarding nodes and data forwarding nodes, the system can clearly understand the functions and responsibilities of each device. This classification can help the system to more accurately identify the role of each device, so as to carry out targeted optimization and adjustment. For example, the optimization focus of the audio output node is audio quality, while the network forwarding node needs to pay more attention to transmission stability and delay control. By labeling the communication path between the detection devices according to the functional classification, the system can ensure that the communication between the nodes can be smooth and efficient. For example, the communication between the audio output node and the audio equipment should try to ensure low latency and low packet loss, while the communication between the network forwarding nodes focuses more on bandwidth stability. In this way, the system can avoid excessive communication burdens between irrelevant devices and reduce unnecessary network pressure.

[0078] In this embodiment, the step S2 of judging whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal further includes:

[0079] S21: Based on the historical synchronization configuration, detecting the room layout information provided by the hotel terminal to the user;

[0080] S22: Determine whether the room layout information is the same as the historical synchronization configuration;

[0081] S23: If not, then identify the power supply location in the hotel room based on the historical synchronization configuration, dynamically adjust the location layout of the audio source device in the hotel room based on the power supply location, and recalibrate the synchronization delay of the audio source device through the distance change of the location layout.

[0082] In this embodiment, the system detects the room layout information provided by the hotel terminal to the user based on the historical synchronization configuration, and then the system determines whether the room layout information is the same as the historical synchronization configuration to execute the corresponding steps; for example, when the system determines that the room layout information provided by the hotel terminal to the user is the same as the historical synchronization configuration, the system will consider that the current room layout is consistent with the previously known configuration, and the user's needs do not need to be significantly adjusted or reconfigured. The system will directly call the existing configuration without re-performing complex scanning, analysis or merging steps, which greatly improves the response speed of audio linkage and audio stream transmission, allowing the system to quickly execute audio playback tasks, while continuing to use the configuration to maintain the settings of the audio routing table, ensuring that the audio stream can be smoothly transmitted in the existing network structure, avoiding potential unstable factors in the audio transmission process, and under the same room layout, the audio equipment and audio output requirements are consistent, so the system can maintain the linkage effect of the audio equipment in the room, such as when multiple rooms play the same or different audio content at the same time, it can ensure synchronization and quality; for example, when the system determines that the room layout information provided by the hotel terminal to the user is different from the historical synchronization configuration, the system will consider that the current room layout is inconsistent with the previously known configuration, and the system will be based on the historical synchronization Configuration, identify the power supply position in the hotel guest room, dynamically adjust the position layout of the audio source equipment in the hotel guest room according to the different power supply positions, and recalibrate the synchronization delay of the audio source equipment through the change of the distance of the position layout; by identifying the power supply position and re-arranging the audio source equipment, the system can more accurately calibrate the synchronization delay between different audio equipment, ensure that the audio equipment plays the best synergy effect in the space, and reduce the delay and asynchronism problems caused by position differences. At the same time, by recalibrating the synchronization delay of the audio source equipment, the system can improve the transmission quality of the audio, reduce the delay and echo problems in the audio playback, and thus enhance the user's audio experience, especially in the case of large-scale, multiple rooms playing at the same time, and when the room layout changes, the system can flexibly adapt to different spatial structures and power supply positions without relying on static configuration, improving the system's adaptive ability. This flexibility enables the system to handle the audio needs of different guest rooms, whether it is layout changes, room size adjustments, or changes in other factors. By identifying the power supply position, the system can optimize the efficiency of power use, reasonably arrange the audio equipment, and avoid the equipment in some areas from not being able to play the best effect due to insufficient power. This optimization not only improves the stability of the audio system, but also helps to reduce energy waste.

[0083] In this embodiment, the step S4 of determining whether the audio routing table detects audio delay further includes:

[0084] S41: Based on the transmission process of the audio signal from the audio source device to the target device, mark the audio packet timestamp of the audio signal between various nodes;

[0085] S42: Determine whether the sending time and receiving time of the audio packet timestamp are within the same preset interval;

[0086] S43: If not, then the audio delay of the audio packet timestamp is calculated according to the sending time and the receiving time, the tolerant delay value from the audio source device to the target device is monitored, and according to the tolerant delay value, the delay compensation measures preset by the hotel terminal are triggered, wherein the delay compensation measures specifically include adjusting the routing path of the audio stream, dynamically allocating the bandwidth of the audio stream, and adjusting the buffer of the audio stream.

[0087] In this embodiment, the system marks the audio packet timestamps of the audio signal between each node based on the transmission process of the audio signal from the audio source device to the target device, and then the system determines whether the sending time and the receiving time of the audio packet timestamp are in the same preset interval to execute the corresponding steps; for example, when the system determines that the sending time and the receiving time of the audio packet timestamp are in the same preset interval, the system will consider that the time synchronization of the audio signal during the transmission process is good, the audio packet is not affected by delays or losses during the transmission process, and meets the set time constraints. The system will adjust the allocation of network resources according to the stability of the audio packet transmission to optimize the bandwidth. The system records the timestamp of each audio packet and its sending and receiving time to form a health report of audio transmission, which is convenient for future analysis and tuning. For example, the system can record the stability of each round of audio flow, which is convenient for maintenance personnel to track and diagnose, and ensure that the sending and receiving time of audio packets are synchronized to avoid delays, freezes or incoherent audio playback, thereby improving the user's audio experience, especially when multiple devices are linked or playing over a large range. For example, when the system determines that the sending time and receiving time of the audio packet timestamp are not in the same preset interval, the system will recognize that In order to prevent the abnormal time synchronization of audio signals during transmission, the system will calculate the audio delay of the audio packet timestamp according to the sending time and receiving time, monitor the tolerance delay value from the audio source device to the target device, and trigger the delay compensation measures pre-set by the hotel terminal based on these tolerance delay values. The delay compensation measures specifically include adjusting the routing path of the audio stream, dynamically allocating the bandwidth of the audio stream, and adjusting the buffer of the audio stream. Through real-time delay compensation, the system can ensure the integrity and high-quality output of the audio, avoid the negative impact of excessive delay or packet loss on the audio quality, and each node of the audio stream will be optimized according to the delay compensation strategy to ensure that the playback effect meets the predetermined standards. According to the real-time network conditions, the system can automatically adjust the transmission strategy of the audio stream to avoid the situation where smooth audio transmission cannot be guaranteed due to excessive network load under traditional static configuration. In the scenario where audio is played on multiple audio devices or rooms, the system can optimize the playback synchronization of each device in real time according to the current delay situation to ensure the time consistency of all devices and avoid the problem of audio asynchrony. For users, delay compensation effectively avoids the perception of delay in audio playback, especially in a hotel environment. When the audio systems in multiple rooms need to play synchronously, reducing delay can improve the overall sound effect and bring users a smoother and more natural listening experience.

[0088] In this embodiment, based on the area to be synchronized preset by the hotel terminal, the step S1 of identifying the corresponding room synchronization configuration from the area to be synchronized further includes:

[0089] S11: Based on the audio synchronization area pre-selected by the user through the hotel terminal, identifying at least two corresponding hotel rooms from the audio synchronization area;

[0090] S12: Determine whether the hotel guest room can detect the preset audio linkage;

[0091] S13: If not, check the audio linkage configuration between the hotel rooms, identify the abnormal type of the audio linkage according to the audio linkage configuration, and review the linkage availability of the hotel rooms through the hotel terminal according to the abnormal type, wherein the abnormal type specifically includes device connection abnormality, audio signal transmission abnormality, and network and hardware abnormality.

[0092] In this embodiment, the system identifies at least two corresponding hotel rooms from the audio synchronization areas pre-selected by the user through the hotel terminal, and then the system determines whether these hotel rooms can detect the pre-set audio linkage to execute the corresponding steps; for example, when the system determines that the hotel room can detect the pre-set audio linkage, the system will consider that the audio equipment in the hotel room has been able to successfully establish a linkage relationship with the audio equipment in other rooms or systems, and the network configuration, audio stream transmission path, etc. are in normal working condition, and the system will perform synchronization tests between the audio equipment to ensure that the audio equipment in multiple rooms can be synchronized in real time according to the preset requirements. The first step is to play the audio stream. This step can ensure that the audio equipment in each room is not out of sync during the audio stream transmission process. At the same time, it checks whether the audio stream path from the audio source device to the target audio equipment is unobstructed, verifies whether the transmission delay and bandwidth of the audio stream meet the preset requirements, ensures that there is no audio freeze or packet loss, and if audio equipment in multiple rooms participate in the linkage, the system will dynamically adjust the routing table of the audio stream according to the needs of each room to ensure that the audio stream can be reasonably distributed to each device and maintain audio synchronization; for example, when the system determines that the hotel room does not detect the pre-set audio linkage, the system will assume that the audio equipment in the hotel room cannot establish a linkage relationship with each other. The system will check the audio linkage configurations between hotel rooms, and identify the abnormal types of audio linkage based on these audio linkage configurations. The abnormal types include device connection abnormalities, audio signal transmission abnormalities, and network and hardware abnormalities. According to different abnormal types, the linkage availability of hotel rooms will be reviewed through hotel terminals; the system can identify audio signal transmission abnormalities to help timely discover audio delays, packet loss or sound quality problems, and can automatically adjust audio stream routing, bandwidth allocation and other parameters according to these abnormal types to ensure the quality and stability of the audio stream and avoid a decline in user experience due to unstable audio signal transmission. At the same time, according to different abnormal types, the system can dynamically adjust the audio Linkage configuration, for example, when an abnormal device connection is detected, the system will automatically retry the device connection or prompt the administrator to operate; when an abnormal audio signal transmission is found, the system will adjust the routing and bandwidth allocation of the audio stream. When a network or hardware problem occurs, the system will adjust the audio output quality, or automatically adjust the buffer size and delay compensation according to the change in network quality. It can also automatically detect anomalies and quickly repair them to ensure that the audio equipment in the guest room can resume normal operation as soon as possible, avoiding the user's stay experience being affected by audio linkage problems. Whether it is background music, voice broadcasting, or the playback of entertainment content, the system can ensure the smooth transmission of the audio stream and improve customer satisfaction.

[0093] Reference Figure 2 , is a central control system for synchronously playing audio in multiple rooms of a hotel in one embodiment of the present invention, comprising:

[0094] An identification module 10, for identifying a corresponding room synchronization configuration from the area to be synchronized based on the area to be synchronized preset by the hotel terminal;

[0095] A judgment module 20, used to judge whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal;

[0096] Execution module 30 is used for, if not, applying a preset network scanning protocol according to the room layout information preset by the hotel terminal, obtaining the real-time audio stream of the room synchronization configuration, dynamically merging the audio stream path of the room synchronization configuration according to the real-time audio stream, identifying the audio requirement preset by the room synchronization configuration, and adaptively adjusting the audio routing table of the hotel terminal according to the audio requirement;

[0097] A second determination module 40 is used to determine whether the audio routing table detects an audio delay;

[0098] The second execution module 50 is used to, if detected, construct an audio stream transmission path diagram of the room synchronization configuration based on the network topology preset in the audio routing table, collect the playback delay of the room synchronization configuration according to the synchronization signal of the audio stream transmission path diagram, apply a preset audio test tool to obtain the error data of the playback delay, and dynamically reduce the compression ratio of the audio encoding according to the error data.

[0099] In this embodiment, the identification module 10 identifies the corresponding room synchronization configuration from the audio synchronization areas pre-set by the hotel terminal, and then the judgment module 20 judges whether the room synchronization configuration set by the user matches the historical synchronization configuration pre-recorded by the hotel terminal to execute the corresponding steps; for example, when the system determines that the room synchronization configuration set by the user can match the historical synchronization configuration pre-recorded by the hotel terminal, the system will consider that the audio linkage requirement set by the user is consistent with the existing configuration of the hotel terminal, and the system can directly execute the operation corresponding to the historical configuration to avoid unnecessary reconfiguration or adjustment, thereby improving efficiency and reducing potential errors. The system will confirm that the room synchronization configuration set by the user is completely consistent with the audio stream, device connection, routing table and other information of the historical configuration to ensure that there is no difference, and ensure that the two match by comparing and checking key information such as device ID, audio routing, audio source, room layout, etc. At the same time, once the match is confirmed, the system can directly load the historical synchronization configuration pre-recorded by the hotel terminal, which means that there is no need Regenerate the audio stream path, audio routing table and other contents, and directly apply the parameters in the historical configuration to reduce the time of calculation and adjustment. Before executing the historical configuration, the system can quickly scan the hotel's network environment to check whether the network indicators such as bandwidth and delay meet the requirements of the historical configuration. If the network conditions do not match, the system can optimize the network environment through intelligent adjustment or prompt the user to ensure the smoothness of audio synchronization. For example, when the system determines that the room synchronization configuration set by the user cannot match the historical synchronization configuration pre-recorded by the hotel terminal, the execution module 30 will think that the audio linkage requirement set by the user is inconsistent with the existing configuration of the hotel terminal, and the system cannot directly execute operations that do not correspond to the historical configuration. The system will apply the pre-set network scanning protocol according to the room layout information pre-set by the hotel terminal to obtain the real-time audio stream of the room synchronization configuration, and dynamically merge the audio stream path of the room synchronization configuration based on these real-time audio streams, identify the audio requirements of the room synchronization configuration, and adaptively adjust the audio routing table of the hotel terminal according to the audio requirements.By applying a pre-set network scanning protocol and acquiring real-time audio streams, the system can dynamically identify and adapt to the audio needs of the room. Whether it is increasing or decreasing the number of rooms, adjusting the volume or changing the audio source, the system can make adjustments based on real-time data. Even if the room synchronization configuration set by the user does not match the historical configuration, the system can flexibly respond to different configuration requirements by acquiring audio streams in real time and dynamically merging audio paths, avoiding the limitations of traditional static configurations that cannot be adapted. At the same time, through the analysis of real-time audio streams, the system can identify the audio needs of each room and intelligently merge audio stream paths, which not only improves The transmission efficiency of the audio stream is improved, and the problems of audio stream conflicts or unreasonable paths that may occur in the traditional configuration are avoided. By dynamically merging the audio stream paths and optimizing the audio routing table, the audio quality and playback synchronization can be effectively guaranteed, and the problems of sound quality degradation and excessive delay that may occur in the traditional system are avoided. Then the second judgment module 40 determines whether the audio routing table detects the audio delay to execute the corresponding steps. For example, when the system determines that the audio routing table of the hotel terminal does not detect the audio delay, the system will consider that the current audio stream transmission is smooth, and the audio synchronization effect of each room of the sound system meets the expected During the period, there is no audio delay caused by network transmission problems or improper device configuration. The system will record the current audio routing table configuration and the transmission status of the audio stream to provide reference data for subsequent room synchronization configuration. If the user makes a new configuration or modification in the future, the system can quickly load the historical configuration to ensure that the new configuration still meets the synchronization requirements. At the same time, since there is no audio delay problem, the system does not need to adjust the compression ratio of the audio encoding, which can avoid affecting the sound quality due to excessive or insufficient compression. In order to ensure the long-term stability of the system, the system should continue to monitor the transmission quality of the audio stream and detect in real time whether there is a new delay or audio quality degradation. Through continuous monitoring, the system can respond to any possible problems in a timely manner; for example, when the system determines that the audio routing table of the hotel terminal can detect audio delay, the second execution module 50 will think that the current audio stream transmission is not smooth. The system will build an audio stream transmission path diagram of the room synchronization configuration based on the network topology pre-set by the audio routing table, collect the playback delay of the room synchronization configuration according to the synchronization signal of the audio stream transmission path diagram, apply the pre-set audio test tool, obtain the error data of the playback delay, and dynamically reduce the compression ratio of the audio encoding based on these error data;By constructing an audio stream transmission path diagram and optimizing the path of the audio stream according to the diagram, the system can find out the network bottleneck or device blockage that causes audio delay. After dynamically adjusting the audio stream path, it can reduce conflicts and blockages during transmission and make the audio signal smoother. At the same time, through synchronous signal acquisition and playback delay monitoring, the system can accurately understand the audio playback delay in each room, and judge whether there is a link that needs to be optimized through data feedback. The collected real-time data can help the system make adjustments at critical moments to improve the quality of the audio stream. In addition, according to the error data, the system can dynamically reduce the compression ratio of audio encoding. By reducing the compression ratio, the system can reduce the distortion that may be introduced during the compression process, especially the sound quality loss under high compression conditions, so as to ensure the quality and clarity of the audio, especially in poor network environments or limited bandwidth. ;

[0100] In this embodiment, it also includes:

[0101] An acquisition module, used for acquiring an audio buffer preset by the hotel terminal based on an audio source device synchronously configured in the room;

[0102] A third judgment module is used to judge whether the audio buffer area conforms to the current network status of the hotel terminal;

[0103] The third execution module is used to identify the change of network quality of the hotel terminal if not, dynamically adjust the audio cache of the audio buffer according to the change of network quality, and adaptively adjust the audio output content of the hotel terminal according to the preset environmental conditions of the room synchronization configuration, wherein the network quality change specifically includes packet loss, delay and jitter, the environmental conditions specifically include noise level, population density and room size, and the audio output content specifically includes audio source content, audio output quality and ambient sound perception.

[0104] In this embodiment, the system obtains the audio buffer pre-set in the hotel terminal based on the audio source device synchronously configured in the room, and then the system determines whether the audio buffer meets the current network status of the hotel terminal to execute the corresponding steps; for example, when the system determines that the audio buffer pre-set in the hotel terminal can meet the current network status of the hotel terminal, the system will believe that the network can smoothly carry the transmission of the current audio stream, and the capacity and performance of the audio buffer are sufficient to cope with the stable and smooth transmission of audio data. The system will continue to use the current audio routing configuration, network bandwidth and buffer size without adjusting the audio stream path, compression rate or other network parameters, which reduces system resource consumption and ensures the high efficiency of audio playback. At the same time, the adaptability of the buffer ensures the smooth transmission of audio, and no additional adjustment or processing is required to solve the problems caused by the network. The audio effect can be consistent and maintain high quality. And if the audio buffer matches the network status, the system can confirm that the transmission path of the audio stream and the room audio linkage mechanism have stabilized, and the audio data is synchronously transmitted between the rooms without delay. The system can continue to play the audio content according to the preset room layout without the need for other complex manual adjustments; for example, when the system determines that the audio buffer pre-set in the hotel terminal cannot meet the current network status of the hotel terminal, the system will consider that the network cannot carry the transmission of the current audio stream. The system will identify the changes in the network quality of the hotel terminal, which specifically include packet loss, delay and jitter. According to the changes in the network quality, the audio cache of the audio buffer is dynamically adjusted, and the audio output content of the hotel terminal is adaptively adjusted according to the environmental conditions pre-set in the room synchronization configuration, which specifically include noise level, population density and room size. The audio output content specifically includes audio source content, audio output quality and ambient sound perception;The system can optimize the audio buffer in a targeted manner through real-time detection of packet loss, delay and jitter, thereby reducing audio interruption, freeze and loss caused by network problems. This real-time network optimization can ensure the complete transmission and seamless playback of audio content. At the same time, it can dynamically adjust the audio buffer according to the real-time network quality, and can also adaptively adjust the audio output content in combination with the environmental conditions of the room. This highly intelligent adjustment method can ensure that even if the network conditions or environmental conditions change, the sound effect is always optimal. By dynamically adjusting the audio settings according to the real-time network conditions and room environment changes, the system can avoid overusing audio resources or unnecessary waste of resources. For example, when the network conditions are good, the system can reduce the size of the buffer and reduce the complexity of audio encoding. When the network is unstable, the system can increase the buffer and reduce the compression ratio to improve the stability of the audio stream. This dynamic adjustment mechanism reduces unnecessary operations and resource consumption. Especially when the network quality is unstable, the system can automatically optimize the transmission path and buffer mechanism of the audio stream to avoid excessive resource occupation or overload of processing capacity. ;

[0105] In this embodiment, the execution module further includes:

[0106] An identification unit, configured to identify the audio format of the room synchronization configuration based on the audio type preset by the user through the hotel terminal, wherein the audio type specifically includes background music, voice broadcast and entertainment content;

[0107] A judging unit, configured to judge whether the audio format matches the audio requirement;

[0108] The execution unit is used to divide the room synchronization configuration into various audio playback areas according to the room layout information, and coordinate the playback strategy of the audio routing table according to the audio playback area, wherein the playback strategy specifically includes single loop, random playback and timed playback.

[0109] In this embodiment, the system identifies the audio format of the room synchronization configuration based on the audio type pre-set by the user through the hotel terminal, the audio type specifically includes background music, voice broadcast and entertainment content, and then the system determines whether these audio formats match the audio type preset in the hotel terminal to execute the corresponding steps; for example, when the system determines that the audio format can match the audio type preset in the hotel terminal, the system will consider that the identified audio stream format is consistent with the audio requirement preset by the system and meets the usage scenario of the current room and the requirements of the user. The system will route the audio signal to each audio output device (such as a speaker or a TV, etc.) according to the current room synchronization configuration. For example, for background music, the system can ensure that the audio signal is synchronized by area Distribution, adapt to the playback needs of different rooms or areas. For voice broadcasts, the audio stream can be transmitted to the room that needs to be broadcast first, and the clarity and delay are minimized. For entertainment content, the system can adjust the loudness and effect of the audio output according to the room size and equipment configuration to provide an immersive experience, and monitor the transmission status of the audio stream in the network in real time to ensure that there are no delays, packet loss or distortion. If network delays or quality problems are detected, the system can automatically perform buffering, compression ratio adjustment or audio stream redirection to ensure audio stability and quality; for example, when the system determines that the audio format of the room synchronization configuration cannot match the audio type preset on the hotel terminal, the system will consider the recognized audio stream format and If the preset audio requirements are inconsistent, the system will divide the room synchronization configuration into various audio playback areas according to the room layout information, and coordinate the playback strategy of the audio routing table according to different audio playback areas. The playback strategy specifically includes single song loop, random playback and timed playback; the system can set different playback strategies for different areas by dividing the room synchronization configuration into multiple audio playback areas. This regionalized control method allows the system to achieve flexible audio playback and management in different rooms or areas even if the audio format cannot completely match the preset audio type. At the same time, by adjusting the playback strategy and regional division, the system can avoid playback conflicts or instability caused by inconsistent audio formats. Different rooms use Different playback strategies can reduce the problems caused by audio stream asynchrony or incompatible audio formats. Through flexible playback strategies, the system can automatically adjust the playback mode of audio content according to the needs of different areas of the hotel to ensure that the audio content is highly matched with the scene. For example, during meal times, the audio system in the restaurant area can play light music or background music, while the conference room may play news or broadcasts. Such intelligent playback control greatly improves the user experience. By dynamically adjusting audio playback according to playback strategies and regional needs, the system can automatically shut down audio equipment or switch to low power consumption mode when needed to avoid energy waste. For example, during non-peak hours, audio equipment can choose timed playback mode to automatically stop playback and reduce energy consumption.

[0110] In this embodiment, the second execution module further includes:

[0111] A collection unit, configured to collect connection relationship paths of device nodes based on device nodes participating in audio stream transmission, wherein the device nodes specifically include audio equipment, routers and switches, and the connection relationship paths specifically include directly connected devices, network segment distribution and bandwidth information;

[0112] A second judgment unit is used to judge whether the connection relationship path can meet the bandwidth requirement;

[0113] The second execution unit is used to, if possible, layer the device nodes according to a preset network structure hierarchy, divide the function classification labels of the device nodes, and detect the communication path between devices based on the function classification labels, wherein the network structure hierarchy specifically includes a core layer, an access layer and an edge layer, and the function classification labels specifically include an audio output node, a network forwarding node and a data forwarding node.

[0114] In this embodiment, the system is based on the device nodes involved in the audio stream transmission, the device nodes specifically include audio equipment, routers and switches, and collects the connection relationship paths of these device nodes, the connection relationship paths specifically include directly connected devices, network segment distribution and bandwidth information, and then the system determines whether these connection relationship paths can meet the bandwidth requirements to execute the corresponding steps; for example, when the system determines that the connection relationship path to the device node cannot meet the bandwidth requirements, the system will believe that the current network resources are insufficient to support high-quality audio stream transmission, which may cause problems such as audio stream freezes, delays or packet loss, thereby affecting the normal operation of the audio equipment. The system will identify the location of the current bandwidth bottleneck by analyzing the connection relationship between the device nodes. The bottleneck nodes are usually devices with insufficient network transmission capabilities (such as routers, switches or certain audio equipment), and these nodes need to be optimized. At the same time, the bandwidth allocation is dynamically adjusted according to priority and usage requirements. For example, priority is given to ensuring the transmission bandwidth of audio streams, limiting the bandwidth usage of non-critical traffic (such as controlling the bandwidth of web browsing or video traffic), and if the network bandwidth is limited, the system can appropriately adjust the compression ratio of the audio stream to reduce bandwidth usage, such as using lower audio quality settings or adjusting the encoding method to reduce the bandwidth required for transmission; for example, when the system determines that the connection path to the device node can meet the bandwidth requirements, the system will then believe that the current network resources are sufficient to support high-quality audio stream transmission, and the system will classify these device nodes according to the pre-set network structure hierarchy, which specifically includes the core layer, access layer, and edge layer, and divide the device nodes into functional categories and labels. The functional classification labels are specific Including audio output nodes, network forwarding nodes and data forwarding nodes, according to different functional classification labels, the communication path between devices is detected; the system can clearly define each part of the network by layering the devices according to the core layer, access layer and edge layer, and formulate corresponding bandwidth requirements, delay restrictions and fault tolerance mechanisms for each layer. This hierarchical structure helps to improve the management efficiency of the network and ensure the coordination and cooperation between the layers. At the same time, by marking the device nodes as audio output nodes, network forwarding nodes and data forwarding nodes, the system can clearly understand the functions and responsibilities of each device. This classification can help the system to more accurately identify the role of each device, so as to carry out targeted optimization and adjustment. For example, the optimization focus of the audio output node is audio quality, while the network forwarding node needs to pay more attention to transmission stability and delay control. By labeling the communication path between the detection devices according to the functional classification, the system can ensure that the communication between the nodes can be smooth and efficient. For example, the communication between the audio output node and the audio equipment should try to ensure low latency and low packet loss, while the communication between the network forwarding nodes focuses more on bandwidth stability. In this way, the system can avoid excessive communication burdens between irrelevant devices and reduce unnecessary network pressure.

[0115] In this embodiment, the judgment module further includes:

[0116] A detection unit, configured to detect the room layout information provided by the hotel terminal to the user based on the historical synchronization configuration;

[0117] A third judgment unit, used to judge whether the room layout information is the same as the historical synchronization configuration;

[0118] The third execution unit is used to, if not, identify the power supply orientation in the hotel room according to the historical synchronization configuration, dynamically adjust the orientation layout of the sound source device in the hotel room according to the power supply orientation, and recalibrate the synchronization delay of the sound source device through the distance change of the orientation layout.

[0119] In this embodiment, the system detects the room layout information provided by the hotel terminal to the user based on the historical synchronization configuration, and then the system determines whether the room layout information is the same as the historical synchronization configuration to execute the corresponding steps; for example, when the system determines that the room layout information provided by the hotel terminal to the user is the same as the historical synchronization configuration, the system will consider that the current room layout is consistent with the previously known configuration, and the user's needs do not need to be significantly adjusted or reconfigured. The system will directly call the existing configuration without re-performing complex scanning, analysis or merging steps, which greatly improves the response speed of audio linkage and audio stream transmission, allowing the system to quickly execute audio playback tasks, while continuing to use the configuration to maintain the settings of the audio routing table, ensuring that the audio stream can be smoothly transmitted in the existing network structure, avoiding potential unstable factors in the audio transmission process, and under the same room layout, the audio equipment and audio output requirements are consistent, so the system can maintain the linkage effect of the audio equipment in the room, such as when multiple rooms play the same or different audio content at the same time, it can ensure synchronization and quality; for example, when the system determines that the room layout information provided by the hotel terminal to the user is different from the historical synchronization configuration, the system will consider that the current room layout is inconsistent with the previously known configuration, and the system will be based on the historical synchronization Configuration, identify the power supply position in the hotel guest room, dynamically adjust the position layout of the audio source equipment in the hotel guest room according to the different power supply positions, and recalibrate the synchronization delay of the audio source equipment through the change of the distance of the position layout; by identifying the power supply position and re-arranging the audio source equipment, the system can more accurately calibrate the synchronization delay between different audio equipment, ensure that the audio equipment plays the best synergy effect in the space, and reduce the delay and asynchronism problems caused by position differences. At the same time, by recalibrating the synchronization delay of the audio source equipment, the system can improve the transmission quality of the audio, reduce the delay and echo problems in the audio playback, and thus enhance the user's audio experience, especially in the case of large-scale, multiple rooms playing at the same time, and when the room layout changes, the system can flexibly adapt to different spatial structures and power supply positions without relying on static configuration, improving the system's adaptive ability. This flexibility enables the system to handle the audio needs of different guest rooms, whether it is layout changes, room size adjustments, or changes in other factors. By identifying the power supply position, the system can optimize the efficiency of power use, reasonably arrange the audio equipment, and avoid the equipment in some areas from not being able to play the best effect due to insufficient power. This optimization not only improves the stability of the audio system, but also helps to reduce energy waste.

[0120] In this embodiment, the second determination module further includes:

[0121] A marking unit, used to mark the audio packet timestamps of the audio signal between various nodes based on the transmission process of the audio signal from the audio source device to the target device;

[0122] A fourth determination unit, used to determine whether the sending time and the receiving time of the audio packet timestamp are within the same preset interval;

[0123] The fourth execution unit is used to calculate the audio delay of the audio packet timestamp according to the sending time and the receiving time, monitor the tolerant delay value from the audio source device to the target device, and trigger the delay compensation measures preset by the hotel terminal according to the tolerant delay value, wherein the delay compensation measures specifically include adjusting the routing path of the audio stream, dynamically allocating the bandwidth of the audio stream, and adjusting the buffer of the audio stream.

[0124] In this embodiment, the system marks the audio packet timestamps of the audio signal between each node based on the transmission process of the audio signal from the audio source device to the target device, and then the system determines whether the sending time and the receiving time of the audio packet timestamp are in the same preset interval to execute the corresponding steps; for example, when the system determines that the sending time and the receiving time of the audio packet timestamp are in the same preset interval, the system will consider that the time synchronization of the audio signal during the transmission process is good, the audio packet is not affected by delays or losses during the transmission process, and meets the set time constraints. The system will adjust the allocation of network resources according to the stability of the audio packet transmission to optimize the bandwidth. The system records the timestamp of each audio packet and its sending and receiving time to form a health report of audio transmission, which is convenient for future analysis and tuning. For example, the system can record the stability of each round of audio flow, which is convenient for maintenance personnel to track and diagnose, and ensure that the sending and receiving time of audio packets are synchronized to avoid delays, freezes or incoherent audio playback, thereby improving the user's audio experience, especially when multiple devices are linked or playing over a large range. For example, when the system determines that the sending time and receiving time of the audio packet timestamp are not in the same preset interval, the system will recognize that In order to prevent the abnormal time synchronization of audio signals during transmission, the system will calculate the audio delay of the audio packet timestamp according to the sending time and receiving time, monitor the tolerance delay value from the audio source device to the target device, and trigger the delay compensation measures pre-set by the hotel terminal based on these tolerance delay values. The delay compensation measures specifically include adjusting the routing path of the audio stream, dynamically allocating the bandwidth of the audio stream, and adjusting the buffer of the audio stream. Through real-time delay compensation, the system can ensure the integrity and high-quality output of the audio, avoid the negative impact of excessive delay or packet loss on the audio quality, and each node of the audio stream will be optimized according to the delay compensation strategy to ensure that the playback effect meets the predetermined standards. According to the real-time network conditions, the system can automatically adjust the transmission strategy of the audio stream to avoid the situation where smooth audio transmission cannot be guaranteed due to excessive network load under traditional static configuration. In the scenario where audio is played on multiple audio devices or rooms, the system can optimize the playback synchronization of each device in real time according to the current delay situation to ensure the time consistency of all devices and avoid the problem of audio asynchrony. For users, delay compensation effectively avoids the perception of delay in audio playback, especially in a hotel environment. When the audio systems in multiple rooms need to play synchronously, reducing delay can improve the overall sound effect and bring users a smoother and more natural listening experience.

[0125] In this embodiment, the identification module further includes:

[0126] A second identification unit, configured to identify corresponding at least two hotel rooms from the audio synchronization area based on the audio synchronization area pre-selected by the user through the hotel terminal;

[0127] A fifth determination unit, used to determine whether the hotel guest room can detect a preset audio linkage;

[0128] The fifth execution unit is used to check the audio linkage configuration between the hotel rooms if not, identify the abnormal type of the audio linkage according to the audio linkage configuration, and review the linkage availability of the hotel rooms through the hotel terminal according to the abnormal type, wherein the abnormal type specifically includes device connection abnormality, audio signal transmission abnormality, and network and hardware abnormality.

[0129] In this embodiment, the system identifies at least two corresponding hotel rooms from the audio synchronization areas pre-selected by the user through the hotel terminal, and then the system determines whether these hotel rooms can detect the pre-set audio linkage to execute the corresponding steps; for example, when the system determines that the hotel room can detect the pre-set audio linkage, the system will consider that the audio equipment in the hotel room has been able to successfully establish a linkage relationship with the audio equipment in other rooms or systems, and the network configuration, audio stream transmission path, etc. are in normal working condition, and the system will perform synchronization tests between the audio equipment to ensure that the audio equipment in multiple rooms can be synchronized in real time according to the preset requirements. The first step is to play the audio stream. This step can ensure that the audio equipment in each room is not out of sync during the audio stream transmission process. At the same time, it checks whether the audio stream path from the audio source device to the target audio equipment is unobstructed, verifies whether the transmission delay and bandwidth of the audio stream meet the preset requirements, ensures that there is no audio freeze or packet loss, and if audio equipment in multiple rooms participate in the linkage, the system will dynamically adjust the routing table of the audio stream according to the needs of each room to ensure that the audio stream can be reasonably distributed to each device and maintain audio synchronization; for example, when the system determines that the hotel room does not detect the pre-set audio linkage, the system will assume that the audio equipment in the hotel room cannot establish a linkage relationship with each other. The system will check the audio linkage configurations between hotel rooms, and identify the abnormal types of audio linkage based on these audio linkage configurations. The abnormal types include device connection abnormalities, audio signal transmission abnormalities, and network and hardware abnormalities. According to different abnormal types, the linkage availability of hotel rooms will be reviewed through hotel terminals; the system can identify audio signal transmission abnormalities to help timely discover audio delays, packet loss or sound quality problems, and can automatically adjust audio stream routing, bandwidth allocation and other parameters according to these abnormal types to ensure the quality and stability of the audio stream and avoid a decline in user experience due to unstable audio signal transmission. At the same time, according to different abnormal types, the system can dynamically adjust the audio Linkage configuration, for example, when an abnormal device connection is detected, the system will automatically retry the device connection or prompt the administrator to operate; when an abnormal audio signal transmission is found, the system will adjust the routing and bandwidth allocation of the audio stream. When a network or hardware problem occurs, the system will adjust the audio output quality, or automatically adjust the buffer size and delay compensation according to the change in network quality. It can also automatically detect anomalies and quickly repair them to ensure that the audio equipment in the guest room can resume normal operation as soon as possible, avoiding the user's stay experience being affected by audio linkage problems. Whether it is background music, voice broadcasting, or the playback of entertainment content, the system can ensure the smooth transmission of the audio stream and improve customer satisfaction.

[0130] Although embodiments of the present invention have been shown and described, it will be appreciated by those skilled in the art that various changes, modifications, substitutions and variations may be made to the embodiments without departing from the principles and spirit of the present invention, and that the scope of the present invention is defined by the appended claims and their equivalents.

Claims

1. A central control method for synchronously playing audio in multiple rooms of a hotel, characterized in that: The following steps are involved: Based on the to-be-synchronized area preset by the hotel terminal, identifying the corresponding room synchronization configuration from the to-be-synchronized area; Determining whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal; If yes, then execute the operation corresponding to the historical synchronization configuration; If not, then according to the preset room layout information of the hotel terminal, apply the preset network scanning protocol to obtain the real-time audio stream of the room synchronization configuration, dynamically merge the audio stream path of the room synchronization configuration according to the real-time audio stream, identify the audio requirements preset in the room synchronization configuration, and adaptively adjust the audio routing table of the hotel terminal according to the audio requirements; determining whether the audio routing table detects an audio delay; If detected, then based on the network topology preset in the audio routing table, an audio stream transmission path diagram of the room synchronization configuration is constructed, and according to the synchronization signal of the audio stream transmission path diagram, the playback delay of the room synchronization configuration is collected, and a preset audio test tool is applied to obtain error data of the playback delay, and according to the error data, the compression ratio of the audio encoding is dynamically reduced; Wherein, the step of identifying the corresponding room synchronization configuration from the area to be synchronized based on the area to be synchronized preset by the hotel terminal further includes: Based on the audio synchronization area pre-selected by the user through the hotel terminal, identifying at least two corresponding hotel rooms from the audio synchronization area; Determining whether the hotel guest room can detect a preset audio linkage; If yes, then conduct synchronization test between audio equipment to ensure that audio equipment in multiple rooms can play audio streams synchronously in real time according to preset requirements; If not, check the audio linkage configuration between the hotel rooms, identify the abnormal type of the audio linkage according to the audio linkage configuration, and review the linkage availability of the hotel rooms through the hotel terminal based on the abnormal type, wherein the abnormal type specifically includes device connection abnormality, audio signal transmission abnormality, and network and hardware abnormality.

2. The central control method for synchronously playing audio in multiple rooms of a hotel according to claim 1, characterized in that: Before the step of applying a preset network scanning protocol to obtain the real-time audio stream of the room synchronization configuration, the step further includes: Based on the audio source device synchronously configured in the room, obtaining the audio buffer preset by the hotel terminal; Determining whether the audio buffer is consistent with the current network status of the hotel terminal; If not, identify the change in network quality of the hotel terminal, dynamically adjust the audio cache of the audio buffer according to the change in network quality, and adaptively adjust the audio output content of the hotel terminal according to the environmental conditions preset in the room synchronization configuration, wherein the network quality change specifically includes packet loss, delay and jitter, the environmental conditions specifically include noise level, population density and room size, and the audio output content specifically includes audio source content, audio output quality and ambient sound perception.

3. The central control method for synchronously playing audio in multiple rooms of a hotel according to claim 1, characterized in that: The step of constructing the audio stream transmission path diagram of the room synchronization configuration based on the network topology preset in the audio routing table further includes: Based on the device nodes involved in the audio stream transmission, the connection relationship paths of the device nodes are collected, wherein the device nodes specifically include audio equipment, routers and switches, and the connection relationship paths specifically include directly connected devices, network segment distribution and bandwidth information; Determine whether the connection relationship path can meet the bandwidth requirement; If possible, the device nodes are layered according to a preset network structure hierarchy, and the functional classification labels of the device nodes are divided. Based on the functional classification labels, the communication paths between devices are detected, wherein the network structure hierarchy specifically includes a core layer, an access layer, and an edge layer, and the functional classification labels specifically include an audio output node, a network forwarding node, and a data forwarding node.

4. The central control method for synchronously playing audio in multiple rooms of a hotel according to claim 1, characterized in that: The step of determining whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal also includes: Based on the historical synchronization configuration, detecting the room layout information provided by the hotel terminal to the user; Determining whether the room layout information is the same as the historical synchronization configuration; If not, the power supply location in the hotel room is identified based on the historical synchronization configuration, and the location layout of the sound source device in the hotel room is dynamically adjusted based on the power supply location. The synchronization delay of the sound source device is recalibrated through the distance change of the location layout.

5. The central control method for synchronously playing audio in multiple rooms of a hotel according to claim 1, characterized in that: The step of determining whether the audio routing table detects audio delay also includes: Based on the transmission process of the audio signal from the audio source device to the target device, marking the audio packet timestamps of the audio signal between various nodes; Determining whether the sending time and the receiving time of the audio packet timestamp are within the same preset interval; If not, then the audio delay of the audio packet timestamp is calculated according to the sending time and the receiving time, the tolerant delay value from the audio source device to the target device is monitored, and based on the tolerant delay value, the delay compensation measures preset by the hotel terminal are triggered, wherein the delay compensation measures specifically include adjusting the routing path of the audio stream, dynamically allocating the bandwidth of the audio stream, and adjusting the buffer of the audio stream.

6. A central control system for synchronously playing audio in multiple rooms of a hotel, characterized in that: include: An identification module, used for identifying a corresponding room synchronization configuration from the area to be synchronized based on the area to be synchronized preset by the hotel terminal; A judgment module, used to judge whether the room synchronization configuration matches the historical synchronization configuration pre-recorded by the hotel terminal; An execution module, configured to, if yes, execute an operation corresponding to the historical synchronization configuration; if no, obtain the real-time audio stream of the room synchronization configuration according to the preset room layout information of the hotel terminal, apply the preset network scanning protocol, dynamically merge the audio stream path of the room synchronization configuration according to the real-time audio stream, identify the audio requirement preset in the room synchronization configuration, and adaptively adjust the audio routing table of the hotel terminal according to the audio requirement; A second determination module is used to determine whether the audio routing table detects audio delay; A second execution module is configured to, if detected, construct an audio stream transmission path diagram of the room synchronization configuration based on the network topology preset in the audio routing table, collect the playback delay of the room synchronization configuration according to the synchronization signal of the audio stream transmission path diagram, apply a preset audio test tool to obtain error data of the playback delay, and dynamically reduce the compression ratio of the audio encoding according to the error data; The identification module also includes: A second identification unit, configured to identify corresponding at least two hotel rooms from the audio synchronization area based on the audio synchronization area pre-selected by the user through the hotel terminal; A fifth determination unit, used to determine whether the hotel guest room can detect a preset audio linkage; The fifth execution unit is used to, if yes, perform synchronization test between audio devices to ensure that the audio devices in multiple rooms can play audio streams synchronously in real time according to preset requirements; if not, check the audio linkage configuration between the hotel rooms, identify the abnormal type of the audio linkage according to the audio linkage configuration, and review the linkage availability of the hotel rooms through the hotel terminal based on the abnormal type, wherein the abnormal type specifically includes device connection abnormality, audio signal transmission abnormality, and network and hardware abnormality.

7. The central control system for synchronously playing audio in multiple rooms of a hotel according to claim 6, characterized in that: Also includes: An acquisition module, used for acquiring an audio buffer preset by the hotel terminal based on the audio source device synchronously configured in the room; A third judgment module is used to judge whether the audio buffer area conforms to the current network status of the hotel terminal; The third execution module is used to identify the change of network quality of the hotel terminal if not, dynamically adjust the audio cache of the audio buffer according to the change of network quality, and adaptively adjust the audio output content of the hotel terminal according to the preset environmental conditions of the room synchronization configuration, wherein the network quality change specifically includes packet loss, delay and jitter, the environmental conditions specifically include noise level, population density and room size, and the audio output content specifically includes audio source content, audio output quality and ambient sound perception.

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