Real-time interaction optimization method and device for voice room application in weak network environment

By using real-time network assessment and dynamic rendering priority adjustment, the issues of lag, latency, and freezing in voice room applications under weak network conditions were resolved, achieving efficient voice interaction and status synchronization, and improving the user experience.

CN121887785APending Publication Date: 2026-04-17GUANGZHOU XINYU NETWORK TECH CO LTD
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Patent Information

Authority / Receiving Office
CN · China
Patent Type
Applications(China)
Current Assignee / Owner
GUANGZHOU XINYU NETWORK TECH CO LTD
Filing Date
2025-12-31
Publication Date
2026-04-17

AI Technical Summary

Technical Problem

In weak network environments, voice room applications suffer from issues such as voice stuttering and latency, inconsistent status synchronization, sluggish operation feedback, and UI freezing, which affect the user's real-time interactive experience.

Method used

By assessing network quality in real time, the rendering priority of UI elements is dynamically adjusted. Frame-level scheduling and layered rendering strategies are adopted to prioritize the transmission of core content. The UI status is updated locally in real time under weak network conditions to provide progressive feedback. Combined with intelligent retry strategies, microphone preemption operations are optimized.

Benefits of technology

In weak network environments, voice latency is reduced to below 800ms, core function availability reaches 95%, operation response latency is reduced by 80%, state synchronization failure rate is reduced to below 5%, and the UI is smooth and freeze-free, providing an adaptive interactive experience.

✦ Generated by Eureka AI based on patent content.

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Abstract

The invention provides a real-time interaction optimization method and device for voice room application in a weak network environment, and relates to the technical field of voice interaction. The method comprises the following steps: calculating a network quality score of a current network environment based on network parameters collected in real time; when a weak network state is judged based on the network quality score, the current rendering priority of the UI element is dynamically adjusted, a rendering task is executed through frame level scheduling and a layered rendering strategy, and the basic rendering priority of the UI element is divided based on a content real-time performance, function importance and user attention three-dimensional evaluation model; when a user initiates a microphone position preemption operation request in a weak network state, the UI state is locally updated in real time, progressive feedback is provided, the microphone position preemption operation request is added into a to-be-confirmed operation queue, and when a microphone grabbing result returned by a server side is received, the UI state is synchronized based on the microphone grabbing result. Rendering and interaction strategies are dynamically adjusted, transmission and display of core content are preferentially guaranteed, and interaction experience in a weak network environment is improved.
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Description

Technical Field

[0001] This invention relates to the field of voice interaction technology, and in particular to a real-time interaction optimization method and apparatus for voice room applications in weak network environments. Background Technology

[0002] With the widespread adoption of mobile internet, voice chat applications (such as voice chat, online karaoke, and gaming) have become a mainstream form of social interaction, with the core requirement being the smoothness of real-time interaction among multiple users. However, in actual usage scenarios, users often face weak network environments (such as subways, remote areas, and network congestion), resulting in high packet loss rates, high latency, and significant jitter, posing a serious challenge to voice chat applications.

[0003] In existing technologies, voice chat applications generally employ a unified loading and synchronization strategy for interactive processing. This means that all content (voice stream, microphone status, gift animations, historical messages, etc.) is processed in a "first-in, first-out" (FIFO) manner, and the client does not update the interface until the server returns data. This approach has significant drawbacks:

[0004] 1. Voice stuttering and latency: In weak network environments (packet loss rate >15%), voice data transmission is obstructed, and latency often exceeds 800ms, which seriously affects real-time communication;

[0005] 2. Inconsistent status synchronization: The synchronization of cross-client status such as microphone position status, microphone entry and exit status, and gift animations relies on unified push from the server. The synchronization failure rate is as high as 35% under weak network conditions, resulting in inconsistent interface statuses seen by different users.

[0006] 3. Sluggish operation feedback: The user operation response is delayed by more than 2 seconds, causing the user to repeat the operation.

[0007] 4. UI freezing phenomenon: The lag in synchronization between the server and client states leads to inconsistent UI display, resulting in a frozen or stuck visual experience. Summary of the Invention

[0008] This invention provides a real-time interaction optimization method and apparatus for voice room applications in weak network environments, which solves the defects of existing voice room applications in weak network environments, such as voice stuttering and delay, inconsistent state synchronization, sluggish operation feedback, and UI freezing. It realizes dynamic adjustment of rendering and interaction strategies, prioritizes the transmission and display of core content, eliminates operation delays, and improves the interactive experience in weak network environments.

[0009] This invention provides a real-time interaction optimization method for voice room applications in weak network environments, applied to the client side, including:

[0010] The network quality score of the current network environment is calculated based on the network parameters collected in real time, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters;

[0011] When the network quality score indicates a weak network state, the current rendering priority of the UI elements is dynamically adjusted, and the rendering task is executed through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention.

[0012] When a user initiates a microphone grabbing request under weak network conditions, the local UI status is updated in real time and progressive feedback is provided. The microphone grabbing request is added to the pending confirmation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

[0013] According to the present invention, a real-time interaction optimization method for voice room applications in a weak network environment includes the following basic rendering priorities:

[0014] Level 1 includes: voice waveform, microphone position status changes, and system warnings;

[0015] The second level includes: gift effect core frames and user speech indicators;

[0016] The third level includes: user avatar updates and regular messages;

[0017] Level 4 includes: historical messages and background animations;

[0018] Level 5 includes: preloaded content;

[0019] The UI elements at the first level will not be downgraded under any network quality score; the UI elements at the second, third, fourth, and fifth levels will be upgraded or downgraded based on the network quality score.

[0020] According to the present invention, a real-time interaction optimization method for voice room applications in a weak network environment is provided, which executes rendering tasks through frame-level scheduling, including:

[0021] At the start of a frame, record the rendering time budget for the current frame;

[0022] According to the adjusted current rendering priority, extract the rendering tasks from high to low;

[0023] For each rendering task, estimate the corresponding rendering time and determine whether the remaining time is sufficient.

[0024] If sufficient time is available, execute the rendering task, deduct the corresponding time, and continue to extract the next priority rendering task;

[0025] If not enough, stop rendering the current frame and submit the rendering task and subsequent rendering tasks to the next frame for processing;

[0026] At the end of the frame, ensure that the rendering time of the current frame does not exceed the rendering time budget to avoid UI freezing.

[0027] According to the present invention, a real-time interaction optimization method for voice room applications in a weak network environment includes a layered rendering strategy comprising:

[0028] Placeholder elements are displayed in real time, wherein the placeholder elements include at least the user's avatar and microphone position status;

[0029] When the network quality score is lower than the first score value and is determined to be a weak network state, load UI elements with low-definition and simplified resources;

[0030] When the network quality score is not less than the first score value and less than the second score value, and is determined to be a good network condition, load the UI elements of the high-definition complete resource.

[0031] When the network quality score is not less than the second score value, and the network is determined to be in a high-quality state, the UI elements of the dynamically enhanced resources are loaded.

[0032] According to the present invention, a real-time interaction optimization method for voice room applications in a weak network environment includes the following steps: when a user initiates a microphone grabbing request in a weak network environment, the UI state is updated locally in real time and progressive feedback is provided; the microphone grabbing request is added to a pending confirmation queue; and when the microphone grabbing result is received from the server, the UI state is synchronized based on the microphone grabbing result.

[0033] When a user initiates a microphone grabbing request in a weak network state, the target microphone position is immediately changed from idle to grabbed, and the microphone grabbing operation button is grayed out locally.

[0034] Add the microphone grabbing request to the local pending confirmation queue and display a text prompt indicating that the microphone grabbing is in progress.

[0035] When the server returns a successful microphone grabbing result, the target microphone position remains in the grabbed state, the microphone position grabbing operation request is removed from the pending confirmation operation queue, and a text prompt indicating successful microphone grabbing is displayed.

[0036] When the server returns a successful microphone grabbing result, the target microphone position remains idle, the microphone grabbing operation button is made clickable again, and a text prompt indicating microphone grabbing failure is displayed.

[0037] The real-time interaction optimization method for voice room applications in a weak network environment provided by the present invention further includes:

[0038] If a user initiates a microphone grabbing request under weak network conditions, and does not receive a microphone grabbing result from the server within a preset time range, the user will re-initiate the microphone grabbing request according to a preset intelligent retry strategy.

[0039] According to the present invention, a real-time interactive optimization method for voice room applications in a weak network environment is provided, wherein the preset intelligent retry strategy adjusts the retry time interval based on the network quality score calculated in real time.

[0040] This invention also provides a real-time interactive optimization device for voice room applications in weak network environments, comprising:

[0041] The network quality scoring module is used to calculate the network quality score of the current network environment based on real-time collected network parameters, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters;

[0042] The UI element rendering module is used to dynamically adjust the current rendering priority of UI elements when the network quality score determines that the network is in a weak network state. It executes rendering tasks through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is divided based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention.

[0043] The UI status feedback module is used to update the UI status locally and provide progressive feedback when a user initiates a microphone grabbing operation request in a weak network state. The microphone grabbing operation request is added to the pending confirmation operation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

[0044] The present invention also provides an electronic device, including a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the processor executes the program to implement the real-time interactive optimization method for voice room applications in weak network environments as described above.

[0045] The present invention also provides a non-transitory computer-readable storage medium having a computer program stored thereon, which, when executed by a processor, implements the real-time interactive optimization method for voice room applications in weak network environments as described above.

[0046] The present invention provides a real-time interactive optimization method and apparatus for voice room applications in weak network environments, which has the following significant advantages over the prior art:

[0047] 1. Core Experience Guarantee: Through priority scheduling and layered rendering, core content such as voice and microphone status are prioritized in weak network conditions. When the packet loss rate is >15%, the voice latency is ≤800ms, and the availability of core functions is ≥95%.

[0048] 2. Extremely fast operation response: Microphone operation is pre-executed locally, reducing perceived latency from 1200ms to 50ms and reducing the rate of repeated operations by 80%;

[0049] 3. Precise Status Synchronization: Through conflict arbitration and intelligent retry, the failure rate of microphone status and gift animation synchronization under weak network conditions is reduced to below 5%, and cross-client consistency is significantly improved;

[0050] 4. Smooth UI without freezing: Frame-level scheduling and time budget management avoid single-frame rendering timeouts, ensuring a stable UI frame rate of over 55fps without freezing.

[0051] 5. Strong adaptability: It dynamically adapts to different network conditions, providing a matching interactive experience from high-quality networks to extremely weak networks, and has wide versatility. Attached Figure Description

[0052] To more clearly illustrate the technical solutions in this invention or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, the drawings described below are some embodiments of this invention. For those skilled in the art, other drawings can be obtained from these drawings without creative effort.

[0053] Figure 1 This is a flowchart illustrating the real-time interactive optimization method for voice room applications in a weak network environment provided by the present invention.

[0054] Figure 2 This is a schematic diagram of the structure of the real-time interactive optimization device for voice room applications in a weak network environment provided by the present invention.

[0055] Figure 3 This is a schematic diagram of the structure of the electronic device provided by the present invention. Detailed Implementation

[0056] To make the objectives, technical solutions, and advantages of this invention clearer, the technical solutions of this invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some, not all, of the embodiments of this invention. All other embodiments obtained by those skilled in the art based on the embodiments of this invention without creative effort are within the scope of protection of this invention.

[0057] It should be noted that, in the description of the embodiments of the present invention, the terms "comprising," "including," or any other variations thereof are intended to cover non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements includes not only those elements but also other elements not expressly listed, or elements inherent to such a process, method, article, or apparatus. Without further limitations, an element defined by the phrase "comprising one..." does not exclude the presence of other identical elements in the process, method, article, or apparatus that includes said element.

[0058] The terms "first," "second," etc., used in this invention are used to distinguish similar objects, not to describe a specific order or sequence. It should be understood that such data can be interchanged where appropriate so that embodiments of the invention can be implemented in orders other than those illustrated or described herein, and the objects distinguished by "first," "second," etc., are generally of the same class, without limiting the number of objects; for example, a first object can be one or more. Furthermore, "and / or" indicates at least one of the connected objects, and the character " / " generally indicates that the preceding and following objects are in an "or" relationship.

[0059] The following is combined Figures 1 to 3 This invention describes a real-time interactive optimization method and apparatus for voice room applications in a weak network environment.

[0060] like Figure 1 As shown, this invention provides a real-time interaction optimization method for voice chat applications in weak network environments, which can be applied to the client of voice social apps, and specifically includes the following steps:

[0061] Step 110: Calculate the network quality score of the current network environment based on the network parameters collected in real time, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters.

[0062] Specifically, the core of this step is to acquire network status data in real time and quantify and score it, providing a basis for decision-making in subsequent optimization strategies:

[0063] (1) Parameter acquisition: Real-time acquisition of key network parameters, including signal strength, packet loss rate, network latency, and network jitter, at a frequency of once per frame (approximately 16ms); among them, packet loss rate is calculated through data packet transmission and reception statistics, latency is obtained through ping test, and jitter is calculated through the standard deviation of 10 consecutive latency.

[0064] (2) Quality score calculation: A weighted algorithm is used to quantify network quality. The weighted algorithm is designed based on the degree of influence of each parameter on the voice interaction experience. The formula is:

[0065] qualityScore=(1-packetLossRate)×0.6f+(1-min(1f,ntt / 1000f))×0.3f+(1-min(1f,jitter / 200f))×0.1f

[0066] Among them, packet loss rate has the highest weight (0.6f) because packet loss has the greatest impact on the integrity of voice transmission; latency has a weight of 0.3f, and jitter has a weight of 0.1f, which comprehensively reflect network stability; the scoring range is 0.0-1.0, and the higher the score, the better the network quality;

[0067] (3) Status Level Classification: Based on the network quality score, the network status can be divided into 5 levels to adapt to different optimization strategies:

[0068] EXCELLENT (0.8f-1.0f): Excellent network, no degradation;

[0069] GOOD (0.6f-0.8f): Good network, slight optimization required;

[0070] FAIR (0.4f-0.6f): Medium network, moderate degradation;

[0071] POOR (0.2f-0.4f): Weak network, deep degradation;

[0072] BAD(<0.2f): Extremely weak network, minimalist mode.

[0073] Step 120: When the network quality score indicates a weak network state, dynamically adjust the current rendering priority of the UI elements and execute the rendering task through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention.

[0074] Specifically, the core of this step is to optimize the allocation of UI rendering resources, ensuring that core content is displayed first and avoiding UI freezing under weak network conditions:

[0075] (1) Priority division: Based on the three-dimensional evaluation model of content real-time performance, functional importance and user attention, the UI elements of the voice room can be divided into 5 priority levels as the basic rendering priority.

[0076] Level 1 includes: voice waveform, microphone position status changes, and system warnings;

[0077] The second level includes: gift effect core frames and user speech indicators;

[0078] The third level includes: user avatar updates and regular messages;

[0079] Level 4 includes: historical messages and background animations;

[0080] Level 5 includes: preloaded content;

[0081] The UI elements at the first level will not be downgraded under any network quality score; the UI elements at the second, third, fourth, and fifth levels will be upgraded or downgraded based on the network quality score.

[0082] The following table shows an example of rendering priority division and its degradation rules:

[0083]

[0084]

[0085] In some embodiments, a dual-threshold degradation mechanism can be used to dynamically adjust priorities and avoid frequent switching.

[0086] (2) Frame-level scheduling: A frame-level scheduling engine is built based on Android Choreographer to ensure that the rendering of a single frame does not exceed the time budget (usually 16ms). The specific process is as follows:

[0087] At the start of a frame, record the rendering time budget for the current frame;

[0088] According to the adjusted current rendering priority, extract the rendering tasks from high to low;

[0089] For each rendering task, estimate the corresponding rendering time and determine whether the remaining time is sufficient.

[0090] If sufficient time is available, execute the rendering task, deduct the corresponding time, and continue to extract the next priority rendering task;

[0091] If not enough, stop rendering the current frame and submit the rendering task and subsequent rendering tasks to the next frame for processing;

[0092] At the end of the frame, ensure that the rendering time of the current frame does not exceed the rendering time budget to avoid UI freezing.

[0093] (3) Layered rendering strategy: A four-stage rendering pipeline is adopted, and automatic rollback is performed when the network condition deteriorates to ensure that the core functions are available.

[0094] First stage: Instantly display placeholder elements, wherein the placeholder elements include at least the user avatar and microphone status;

[0095] Second stage: When the network quality score is lower than the first score value and is judged as a weak network state, load UI elements with low-definition simplified resources;

[0096] Third stage: When the network quality score is not less than the first score value and less than the second score value, and is judged as a good network condition, load the UI elements of the high-definition complete resource;

[0097] Fourth stage: When the network quality score is not less than the second score value and is determined to be a high-quality network state, load the UI elements of the dynamically enhanced resources.

[0098] Step 130: When a user initiates a microphone grabbing operation request in a weak network state, the local UI status is updated in real time and progressive feedback is provided. The microphone grabbing operation request is added to the pending confirmation operation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

[0099] Specifically, the core of this step is to optimize the interactive response of microphone operation and eliminate operation latency and "black hole" phenomenon under weak network conditions:

[0100] When a user initiates a microphone grabbing request in a weak network state, the target microphone position is changed from idle to grabbed in real time locally, and the microphone grabbing operation button is grayed out, which can reduce the operation perception delay from 1200ms to 50ms.

[0101] The microphone grabbing operation request is added to the local pending confirmation operation queue (using versioned status management, assigning a unique version number to each operation to ensure that the operation is traceable and rollbackable), and a text prompt indicating that the microphone grabbing is in progress is displayed.

[0102] When the server returns a successful microphone grabbing result, the target microphone position remains in the grabbed state, the microphone position grabbing operation request is removed from the pending confirmation operation queue, and a text prompt indicating successful microphone grabbing is displayed.

[0103] When the server returns a successful microphone grabbing result, the target microphone position remains idle, the microphone grabbing operation button is made clickable again, and a text prompt indicating microphone grabbing failure is displayed.

[0104] The UI status uses progressive feedback: the operation progress is displayed through a three-level state machine.

[0105] (1) When an operation is initiated: display immediate feedback (such as graying out the button or loading animation);

[0106] (2) While waiting for confirmation: Display the message "Waiting for response";

[0107] (3) When confirming success / failure: display result prompts (such as "Successfully grabbed the microphone" or "Failed to grab the microphone, it has been occupied"); to avoid users having to repeat the operation due to lack of feedback.

[0108] In some embodiments, the method further includes: after a user initiates the microphone position grabbing operation request in a weak network state, if no microphone grabbing result is received from the server within a preset time range, the user re-initiates the microphone position grabbing operation request according to a preset intelligent retry strategy.

[0109] In this embodiment, the preset intelligent retry strategy adjusts the retry time interval based on the real-time calculated network quality score, and dynamically adjusts the retry strategy according to the network quality to avoid wasting resources on ineffective retries.

[0110] The following is a sample of a preset smart retry policy:

[0111] Network rating > 0.6 (GOOD / EXCELLENT): Retry every 500ms, timeout period 3s;

[0112] 0.3 ≤ score ≤ 0.6 (FAIR / POOR): Retry once every 1000ms, timeout time is 5s;

[0113] Score <0.3 (BAD): Retry every 2000ms, timeout period 8s;

[0114] If the attempt fails after 3 retries, a message will pop up saying "Poor network connection, operation failed," allowing users to manually retry.

[0115] It's important to understand that when multiple users simultaneously initiate conflicting operations such as microphone preemption, the server makes decisions based on a two-factor approach: "logical clock + priority weight."

[0116] Logical clock: Records the time sequence of operations, with earlier operations having higher priority;

[0117] Priority and weighting: Weights are assigned based on user level and the importance of the current scenario, with higher weights for users in key scenarios;

[0118] The arbitration result is synchronized to all clients to ensure consistency.

[0119] It should be noted that this system will continuously monitor changes in network status, upgrade the rendering level and interaction strategy when network quality improves, and trigger graceful degradation when network deteriorates, to ensure a stable interactive experience across all scenarios.

[0120] Status monitoring: Continuously monitor changes in network quality score, with a monitoring frequency of once per frame;

[0121] Policy downgrade: If the network score drops from 0.565 (FAIR level) to 0.35 (POOR level), a deep downgrade is triggered: HIGH level is downgraded to MEDIUM level, MEDIUM level is downgraded to LOW level, and gift effects are downgraded from core frame display to text prompts;

[0122] Strategy upgrade: If the network score rises from 0.35 (POOR level) to 0.85 (EXCELLENT level), a strategy upgrade is triggered: restore full priority, load high-definition avatars and complete gift effects, play dynamic effects normally, and adjust the intelligent retry frequency to once every 500ms.

[0123] State synchronization: When network state changes lead to policy adjustments, the adjusted rendering rules and interaction policies are synchronized to the local cache to ensure that subsequent operations are executed according to the latest policy.

[0124] The above-described method for optimizing real-time interaction in voice chat applications under weak network conditions offers the following significant advantages over existing technologies:

[0125] 1. Core Experience Guarantee: Through priority scheduling and layered rendering, core content such as voice and microphone status are prioritized in weak network conditions. When the packet loss rate is >15%, the voice latency is ≤800ms, and the availability of core functions is ≥95%.

[0126] 2. Extremely fast operation response: Microphone operation is pre-executed locally, reducing perceived latency from 1200ms to 50ms and reducing the rate of repeated operations by 80%;

[0127] 3. Precise Status Synchronization: Through conflict arbitration and intelligent retry, the failure rate of microphone status and gift animation synchronization under weak network conditions is reduced to below 5%, and cross-client consistency is significantly improved;

[0128] 4. Smooth UI without freezing: Frame-level scheduling and time budget management avoid single-frame rendering timeouts, ensuring a stable UI frame rate of over 55fps without freezing.

[0129] 5. Strong adaptability: It dynamically adapts to different network conditions, providing a matching interactive experience from high-quality networks to extremely weak networks, and has wide versatility.

[0130] The following describes the real-time interaction optimization device for voice room applications in a weak network environment provided by the present invention. The real-time interaction optimization device for voice room applications in a weak network environment described below and the real-time interaction optimization method for voice room applications in a weak network environment described above can be referred to in correspondence with each other.

[0131] like Figure 2 As shown, the present invention also provides a real-time interactive optimization device for voice room applications in weak network environments, comprising:

[0132] The network quality scoring module 210 is used to calculate the network quality score of the current network environment based on the network parameters collected in real time. The network parameters include: signal strength, packet loss rate, latency, and jitter parameters.

[0133] The UI element rendering module 220 is used to dynamically adjust the current rendering priority of UI elements when the network quality score determines that the network is in a weak network state. The rendering task is executed through frame-level scheduling and layered rendering strategy. The basic rendering priority of the UI elements is divided based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention.

[0134] The UI status feedback module 230 is used to update the UI status locally and provide progressive feedback when a user initiates a microphone grabbing operation request in a weak network state. The microphone grabbing operation request is added to the pending confirmation operation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

[0135] Figure 3 An example is a schematic diagram of the physical structure of an electronic device, such as... Figure 3 As shown, the electronic device may include: a processor 310, a communications interface 320, a memory 330, and a communication bus 340, wherein the processor 310, the communications interface 320, and the memory 330 communicate with each other via the communication bus 340. The processor 310 can call logical instructions in the memory 330 to execute a real-time interactive optimization method for voice room applications in weak network environments. This method includes:

[0136] The network quality score of the current network environment is calculated based on the network parameters collected in real time, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters;

[0137] When the network quality score indicates a weak network state, the current rendering priority of the UI elements is dynamically adjusted, and the rendering task is executed through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention.

[0138] When a user initiates a microphone grabbing request under weak network conditions, the local UI status is updated in real time and progressive feedback is provided. The microphone grabbing request is added to the pending confirmation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

[0139] Furthermore, the logical instructions in the aforementioned memory 330 can be implemented as software functional units and, when sold or used as independent products, can be stored in a computer-readable storage medium. Based on this understanding, the technical solution of the present invention, essentially, or the part that contributes to the prior art, or a part of the technical solution, can be embodied in the form of a software product. This computer software product is stored in a storage medium and includes several instructions to cause a computer device (which may be a personal computer, server, or network device, etc.) to execute all or part of the steps of the methods described in the various embodiments of the present invention. The aforementioned storage medium includes various media capable of storing program code, such as USB flash drives, portable hard drives, read-only memory (ROM), random access memory (RAM), magnetic disks, or optical disks.

[0140] On the other hand, the present invention also provides a computer program product, which includes a computer program that can be stored on a non-transitory computer-readable storage medium. When the computer program is executed by a processor, the computer is able to execute the real-time interactive optimization method for voice room applications in weak network environments provided by the above methods. The method includes:

[0141] The network quality score of the current network environment is calculated based on the network parameters collected in real time, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters;

[0142] When the network quality score indicates a weak network state, the current rendering priority of the UI elements is dynamically adjusted, and the rendering task is executed through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention.

[0143] When a user initiates a microphone grabbing request under weak network conditions, the local UI status is updated in real time and progressive feedback is provided. The microphone grabbing request is added to the pending confirmation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

[0144] In another aspect, the present invention also provides a non-transitory computer-readable storage medium having a computer program stored thereon, which, when executed by a processor, implements a real-time interactive optimization method for voice room applications in weak network environments provided by the methods described above, the method comprising:

[0145] The network quality score of the current network environment is calculated based on the network parameters collected in real time, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters;

[0146] When the network quality score indicates a weak network state, the current rendering priority of the UI elements is dynamically adjusted, and the rendering task is executed through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention.

[0147] When a user initiates a microphone grabbing request under weak network conditions, the local UI status is updated in real time and progressive feedback is provided. The microphone grabbing request is added to the pending confirmation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

[0148] The device embodiments described above are merely illustrative. The units described as separate components may or may not be physically separate. The components shown as units may or may not be physical units; that is, they may be located in one place or distributed across multiple network units. Some or all of the modules can be selected to achieve the purpose of this embodiment according to actual needs. Those skilled in the art can understand and implement this without any creative effort.

[0149] Through the above description of the embodiments, those skilled in the art can clearly understand that each embodiment can be implemented by means of software plus necessary general-purpose hardware platforms, and of course, it can also be implemented by hardware. Based on this understanding, the above technical solutions, in essence or the part that contributes to the prior art, can be embodied in the form of a software product. This computer software product can be stored in a computer-readable storage medium, such as ROM / RAM, magnetic disk, optical disk, etc., and includes several instructions to cause a computer device (which may be a personal computer, server, or network device, etc.) to execute the methods described in the various embodiments or some parts of the embodiments.

[0150] Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention, and not to limit them; although the present invention has been described in detail with reference to the foregoing embodiments, those skilled in the art should understand that modifications can still be made to the technical solutions described in the foregoing embodiments, or equivalent substitutions can be made to some of the technical features; and these modifications or substitutions do not cause the essence of the corresponding technical solutions to deviate from the spirit and scope of the technical solutions of the embodiments of the present invention.

Claims

1. A method for real-time interaction optimization of voice room applications in a weak network environment, characterized in that, Applied to the client side, including: The network quality score of the current network environment is calculated based on the network parameters collected in real time, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters; When the network quality score indicates a weak network state, the current rendering priority of the UI elements is dynamically adjusted, and the rendering task is executed through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention. When a user initiates a microphone grabbing request under weak network conditions, the local UI status is updated in real time and progressive feedback is provided. The microphone grabbing request is added to the pending confirmation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

2. The method for optimizing real-time interaction of voice room application in weak network environment according to claim 1, characterized in that, The basic rendering priority includes: Level 1 includes: voice waveform, microphone position status changes, and system warnings; The second level includes: gift effect core frames and user speech indicators; The third level includes: user avatar updates and regular messages; Level 4 includes: historical messages and background animations; Level 5 includes: preloaded content; The UI elements at the first level will not be downgraded under any network quality score; the UI elements at the second, third, fourth, and fifth levels will be upgraded or downgraded based on the network quality score.

3. The method for optimizing real-time interaction of voice room application in weak network environment according to claim 1, characterized in that, Rendering tasks are executed via frame-level scheduling, including: At the start of a frame, record the rendering time budget for the current frame; According to the adjusted current rendering priority, extract the rendering tasks from high to low; For each rendering task, estimate the corresponding rendering time and determine whether the remaining time is sufficient. If sufficient time is available, execute the rendering task, deduct the corresponding time, and continue to extract the next priority rendering task; If not enough, stop rendering the current frame and submit the rendering task and subsequent rendering tasks to the next frame for processing; At the end of the frame, ensure that the rendering time of the current frame does not exceed the rendering time budget to avoid UI freezing.

4. The method for optimizing real-time interaction of voice room application in weak network environment according to claim 1, characterized in that, The layered rendering strategy includes: Placeholder elements are displayed in real time, wherein the placeholder elements include at least the user's avatar and microphone position status; When the network quality score is lower than the first score value and is determined to be a weak network state, load UI elements with low-definition and simplified resources; When the network quality score is not less than the first score value and less than the second score value, and is determined to be a good network condition, load the UI elements of the high-definition complete resource. When the network quality score is not less than the second score value, and the network is determined to be in a high-quality state, the UI elements of the dynamically enhanced resources are loaded.

5. The real-time interactive optimization method for voice room applications in a weak network environment according to claim 1, characterized in that, When a user initiates a microphone grabbing request under weak network conditions, the local UI status is updated in real time and progressive feedback is provided. The microphone grabbing request is added to the pending confirmation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result, including: When a user initiates a microphone grabbing request in a weak network state, the target microphone position is immediately changed from idle to grabbed, and the microphone grabbing operation button is grayed out locally. Add the microphone grabbing request to the local pending confirmation queue and display a text prompt indicating that the microphone grabbing is in progress. When the server returns a successful microphone grabbing result, the target microphone position remains in the grabbed state, the microphone position grabbing operation request is removed from the pending confirmation operation queue, and a text prompt indicating successful microphone grabbing is displayed. When the server returns a successful microphone grabbing result, the target microphone position remains idle, the microphone grabbing operation button is made clickable again, and a text prompt indicating microphone grabbing failure is displayed.

6. The real-time interactive optimization method for voice room applications in a weak network environment according to claim 1, characterized in that, Also includes: If a user initiates a microphone grabbing request under weak network conditions, and does not receive a microphone grabbing result from the server within a preset time range, the user will re-initiate the microphone grabbing request according to a preset intelligent retry strategy.

7. The real-time interactive optimization method for voice room applications in a weak network environment according to claim 6, characterized in that, The preset intelligent retry strategy adjusts the retry time interval based on the network quality score calculated in real time.

8. A real-time interactive optimization device for voice room applications in a weak network environment, characterized in that, include: The network quality scoring module is used to calculate the network quality score of the current network environment based on real-time collected network parameters, wherein the network parameters include: signal strength, packet loss rate, latency, and jitter parameters; The UI element rendering module is used to dynamically adjust the current rendering priority of UI elements when the network quality score determines that the network is in a weak network state. It executes rendering tasks through frame-level scheduling and layered rendering strategies. The basic rendering priority of the UI elements is divided based on a three-dimensional evaluation model of content real-time performance, functional importance, and user attention. The UI status feedback module is used to update the UI status locally and provide progressive feedback when a user initiates a microphone grabbing operation request in a weak network state. The microphone grabbing operation request is added to the pending confirmation operation queue. When the microphone grabbing result is received from the server, the UI status is synchronized based on the microphone grabbing result.

9. An electronic device comprising a memory, a processor, and a computer program stored in the memory and executable on the processor, characterized in that, When the processor executes the program, it implements the real-time interactive optimization method for voice room applications in a weak network environment as described in any one of claims 1 to 7.

10. A non-transitory computer-readable storage medium having a computer program stored thereon, characterized in that, When the computer program is executed by the processor, it implements the real-time interactive optimization method for voice room applications in a weak network environment as described in any one of claims 1 to 7.