Audio Parameter Adjustment Method, Device, Apparatus and Storage Medium

Abstract

Embodiments of the present application provide an audio parameter adjustment method, device, apparatus, and storage medium. In the embodiments of the present application, according to the actual delay duration of audio frames from the source end to the destination end within a set time period, and the callback time interval for the destination end to callback audio frames from the audio buffer within this time period, the audio quality parameters of the destination end are determined, realizing the quantification of the audio quality at the destination end. Further, according to the audio quality parameters of the destination end, adjusting the audio stream parameters of the destination end realizes the automatic adjustment of the audio stream parameters at the destination end, which helps to improve the efficiency of audio parameter adjustment. On the other hand, based on the audio quality parameters, adjusting the audio stream parameters of the destination end ensures a certain level of audio quality at the destination end, which helps to guarantee the audio quality at the destination end.

Description

Technical Field

[0001] This application relates to the field of Internet technologies, and in particular, to an audio parameter adjustment method, device, apparatus, and storage medium. Background Art

[0002] With the increasing maturity of cloud computing development, it is a current mainstream development trend to configure program modules originally located in terminal devices on the server side to thin down the terminal devices. Based on this, products such as cloud desktops and cloud applications have emerged.

[0003] The audio in a cloud desktop or cloud application is transmitted over the network, and the audio quality is greatly affected by software and hardware conditions such as network fluctuations and the scheduling of the Central Processing Unit (CPU) of the client. Audio stream parameters affect the audio quality to a certain extent. In existing solutions, the audio stream parameters are generally adjusted manually by an administrator or a user, and it cannot be guaranteed whether the impact of the adjusted audio stream parameters on the audio quality is positive or negative. Generally, multiple adjustments are required, and the adjustment efficiency is low. Summary of the Invention

[0004] Multiple aspects of this application provide an audio parameter adjustment method, device, apparatus, and storage medium to achieve automatic adjustment of audio stream parameters, which helps to improve the adjustment efficiency of audio stream parameters.

[0005] An embodiment of this application provides an audio parameter adjustment method, including:

[0006] Obtain the actual delay duration of audio frames from the source end to the destination end within a set time period;

[0007] Obtain the callback time interval for the destination end to callback audio frames from the audio buffer within the set time period;

[0008] Determine the audio quality parameters of the destination end according to the actual delay duration and the callback time interval;

[0009] Adjust the audio stream parameters of the destination end according to the audio quality parameters.

[0010] Another embodiment of this application provides an audio parameter adjustment method, including:

[0011] Obtain the actual delay duration of audio frames from the server to the client within a set time period; the server and the client are a cloud desktop server and a cloud desktop client, or a cloud application server and a cloud desktop client;

[0012] Obtain the callback time interval for the client to callback audio frames from the audio buffer within the set time period;

[0013] Determine the audio quality parameter of the client according to the actual delay duration and the callback time interval;

[0014] Adjust the audio stream parameter of the client according to the audio quality parameter.

[0015] An embodiment of the present application further provides an audio parameter adjustment device, including:

[0016] An acquisition module, configured to acquire the actual delay duration of an audio frame from a source end to a destination end within a set time period; and acquire the callback time interval for the destination end to callback the audio frame from an audio buffer within the set time period;

[0017] A determination module, configured to determine the audio quality parameter of the destination end according to the actual delay duration and the callback time interval;

[0018] An adjustment module, configured to adjust the audio stream parameter of the destination end according to the audio quality parameter.

[0019] An embodiment of the present application further provides an audio parameter adjustment device, including:

[0020] An acquisition module, configured to acquire the actual delay duration of an audio frame from a server to a client within a set time period; and acquire the callback time interval for the client to callback the audio frame from an audio buffer within the set time period; the server and the client are a cloud desktop server and a cloud desktop client, or a cloud application server and a cloud desktop client;

[0021] A determination module, configured to determine the audio quality parameter of the client according to the actual delay duration and the callback time interval;

[0022] An adjustment module, configured to adjust the audio stream parameter of the client according to the audio quality parameter.

[0023] An embodiment of the present application further provides a computing device, including: a memory and a processor; wherein, the memory is used to store a computer program;

[0024] The processor is coupled to the memory and is configured to execute the computer program to execute the steps in the above various audio quality adjustment methods.

[0025] An embodiment of the present application further provides a computer-readable storage medium storing computer instructions, which, when executed by one or more processors, cause the one or more processors to execute the steps in the above various audio quality adjustment methods.

[0026] In the embodiments of the present application, according to the actual delay duration of the audio frames from the source end to the destination end within a set time period, and the callback time interval for the destination end to callback the audio frames from the audio buffer within this time period, the audio quality parameters of the destination end are determined, realizing the quantification of the audio quality at the destination end. Further, according to the audio quality parameters of the destination end, adjusting the audio stream parameters of the destination end realizes the automatic adjustment of the audio stream parameters at the destination end, which helps to improve the efficiency of audio parameter adjustment. On the other hand, based on the audio quality parameters, adjusting the audio stream parameters of the destination end ensures a certain level of the audio quality at the destination end, which helps to guarantee the audio quality at the destination end. BRIEF DESCRIPTION OF THE DRAWINGS

[0027] The drawings described herein are used to provide a further understanding of the present application and form a part of the present application. The schematic embodiments of the present application and their descriptions are used to explain the present application and do not constitute an improper limitation to the present application. In the drawings:

[0028] Figure 1 and Figure 2 is a schematic structural diagram of the data processing system provided by the embodiments of the present application;

[0029] Figure 3a and Figure 3b is a schematic flowchart of the audio parameter adjustment method provided by the embodiments of the present application;

[0030] Figure 4 is a schematic structural diagram of the computing device provided by the embodiments of the present application;

[0031] Figure 5 is a schematic structural diagram of the audio parameter adjustment device provided by the embodiments of the present application. DETAILED DESCRIPTION OF THE EMBODIMENTS

[0032] To make the objectives, technical solutions and advantages of the present application clearer, the technical solutions of the present application will be clearly and completely described below in conjunction with the specific embodiments of the present application and the corresponding drawings. Obviously, the described embodiments are only a part of the embodiments of the present application, rather than all the embodiments. Based on the embodiments in the present application, all other embodiments obtained by those of ordinary skill in the art without creative efforts shall fall within the protection scope of the present application.

[0033] In existing solutions, the audio stream parameters are generally adjusted manually by an administrator or a user. It is impossible to ensure whether the impact of the adjusted audio stream parameters on the audio quality is positive or negative. Generally, multiple adjustments are required, and the adjustment efficiency is low. To solve this technical problem, in some embodiments of the present application, the audio quality parameters of the destination end can be determined according to the actual delay duration of the audio frames from the source end to the destination end within a set time period, and the callback time interval for the destination end to callback the audio frames from the audio buffer within this time period, realizing the quantification of the audio quality at the destination end. Further, according to the audio quality parameters of the destination end, adjusting the audio stream parameters of the destination end realizes the automatic adjustment of the audio stream parameters of the destination end, which helps to improve the audio parameter adjustment efficiency. On the other hand, adjusting the audio stream parameters of the destination end based on the audio quality parameters provides a certain guarantee for the audio quality of the destination end, which helps to ensure the audio quality of the destination end.

[0034] The following will describe in detail the technical solutions provided by each embodiment of the present application with reference to the accompanying drawings.

[0035] It should be noted that the same reference numerals represent the same object in the following drawings and embodiments. Therefore, once an object is defined in one drawing or embodiment, it does not need to be further discussed in the subsequent drawings and embodiments.

[0036] Figure 1 and Figure 2 is a schematic structural diagram of a data processing system provided by an embodiment of the present application. As Figure 1 and Figure 2 shown, the data processing system may include: a source end 10 and a destination end 20. Among them, the source end refers to the device that provides audio; the destination end is the device that receives and plays audio.

[0037] Depending on the application scenario, the implementation forms of the source end 10 and the destination end 20 are different. In a cloud desktop or cloud application scenario, the source end 10 may be the server of the cloud desktop or cloud application, and the destination end 20 may be the client of the cloud desktop or cloud application. In an audio and video call scenario, the source end and the destination end can be implemented as multiple terminals of the audio and video call. In an audio and video call scenario, the source end and the destination end change with the sound source. For example, in a call, the terminal of the current speaker can be the source end; the terminal of the listener can be the destination end, etc.

[0038] In this embodiment, in the cloud desktop or cloud application scenario, the applications running in the cloud environment are collectively referred to as cloud-side applications. Cloud-side applications refer to the applications deployed in the cloud environment, which can be cloud applications or the applications running in the cloud desktop. Among them, cloud applications are the embodiment of cloud computing technology at the application layer. The working principle of cloud applications changes the traditional software usage mode of "local installation and local operation" to the service of "instant access and use". The client does not need to install the corresponding software and can connect to and control the remote service cluster through the Internet or local area network to complete transaction logic or computing tasks.

[0039] A cloud desktop is based on virtualization technology for computer hardware resources, which makes them virtualized into multiple virtual computers, and the desktop operating system without any modification can be directly run on the virtual computers. At the same time, the cloud desktop also provides an operating system that supports virtualization technology, and virtualized applications can be directly run on top of this desktop operating system. The applications running in the cloud desktop refer to the applications running in the desktop operating system. The applications running in the cloud desktop can be applications with a user interface (UI), such as instant messaging applications, online shopping applications, social applications, and email applications, etc. Of course, the applications running in the cloud desktop can also be applications without a UI, such as background scripts and system services running in the cloud desktop.

[0040] In the embodiment of this application, the client of the cloud desktop or cloud application refers to the electronic device used by the user and having functions such as communication. For example, it can be a mobile phone, a tablet computer, a personal computer, a wearable device, or an Internet of Things (IoT) device, etc. Of course, the client can also be a thin client such as a set-top box. The client usually includes at least one processing unit and at least one memory. The client will also include basic configurations such as a network card chip, an input / output (I / O) bus, audio and video components, and a display component.

[0041] In the embodiment of this application, the server of the cloud-side application refers to the computer device that can provide cloud-side application services and generally has the ability to undertake and guarantee services. The server can respond to the service requests of the client and provide services related to the cloud-side application for users. The server can be a single server device, a cloudified server array, or a virtual machine (VM) running in the cloudified server array. In addition, the server can also refer to other computing devices with corresponding service capabilities, such as terminal devices (running service programs) like computers, etc.

[0042] For a cloud desktop system, application programs of the cloud desktop and other application programs running within the cloud desktop are deployed on the server side. The number of application programs running within the cloud desktop can be one or more. "More than one" means two or more. The multiple application programs can be deployed on the same physical machine or on different physical machines. For a cloud application system, application programs of the cloud application are deployed on the server side. The number of application programs of the cloud application can also be one or more. In the embodiments of the present application, the implementation form of the cloud application is not limited. Optionally, the cloud application can be a web page, client software, an application (APP), or a mini-program, etc.

[0043] In the cloud desktop or cloud application scenario, the server can store audio data of the cloud-side application and send the audio data to the client in response to the request of the client. In an audio-video call scenario, the source end can collect audio data of the source-end user through a pickup and send the audio data to other terminals (i.e., the destination end) in this call. The number of destination ends can be one or more. "More than one" means two or more. Figure 1 and Figure 2 Only the data processing system is taken as an example of the cloud desktop or cloud application system for exemplary illustration, but it does not constitute a limitation.

[0044] For the destination end 20 that receives audio data, due to network jitter, the audio frames may not reach the speaker 202 of the destination end 20 on time, resulting in audio stuttering or even disconnection. To mitigate the phenomenon of audio stuttering caused by network jitter and the like, the destination end 20 can set an audio buffer 201. The audio buffer 201 can be set in the memory of the destination end 20 and is used to store the audio frames sent by the source end 10. The audio buffer 201 can be a first-in-first-out (FIFO) audio buffer queue. The data volume of the audio frames stored in the audio buffer 201 is less than or equal to the capacity of the audio buffer 201. Generally, the data volume of the audio frames stored in the audio buffer 201 is equal to the capacity of the audio buffer 201.

[0045] Based on the audio buffer 201, the destination end 20 can obtain an audio stream from the source end 10, store the audio stream in the audio buffer 201, and then evenly send the audio stream to the speaker 202 of the destination end 20 for playback through a callback function. For example, in the cloud desktop or cloud application scenario, the server can, in response to the request of the client, send the currently requested audio data in the form of an audio stream to the client. The client can store the received audio frames in the audio buffer 201. Then, the client can evenly send the audio stream to the speaker 202 of the client for playback through a callback function.

[0046] Specifically, the audio thread in the destination end 20 obtains the audio frames to be played through a callback function. Specifically, the audio thread in the destination end 20 continuously sends an empty buffer queue to the callback function. The callback function is responsible for filling the buffer queue with audio data and then sending the buffer queue to the playback system of the destination end 20 for playback.

[0047] Based on the above audio buffer 201, as Figure 1 shown, the destination end 20 can obtain the audio frames sent by the source end and store the audio frames sent by the source end in the audio buffer 201. Further, the audio thread of the destination end 20 can obtain the audio frames to be played from the audio buffer through a callback function; and transmit the audio frames to be played to the speaker 202 of the destination end for playback. This process is the local scheduling process of the destination end.

[0048] In the embodiment of the present application, the audio stream quality mainly reflects the smoothness of video playback at the destination end. The inventors of the present application have found through research that the audio stream quality of the destination end 20 is mainly affected by network jitter and audio scheduling jitter at the local end of the destination end. Among them, network jitter, also known as audio transmission jitter, is used to describe the degree of change in the delay of audio from the source end to the destination end. The audio scheduling jitter at the local end of the destination end is used to describe the degree of change in the delay of scheduling audio frames from the audio buffer to the speaker for playback at the destination end.

[0049] Since audio transmission jitter and audio scheduling jitter at the destination end are the main factors affecting the audio stream quality of the destination end, therefore, the audio stream parameters of the destination end can be adjusted according to the audio transmission jitter parameters and audio scheduling jitter parameters. Audio stream parameters refer to the parameters that affect the audio quality of the destination end, specifically, the parameters that affect the audio playback quality of the destination end. In this embodiment, the audio stream parameters can be implemented as parameters that affect audio transmission jitter and audio scheduling jitter at the destination end. In this embodiment, the audio stream quality can be ensured by adjusting the audio stream parameters of the destination end.

[0050] Based on the above analysis, since audio transmission jitter describes the degree of change in the delay of audio from the source end to the destination end, therefore, as Figure 2 shown, in order to determine the audio quality of the destination end, the destination end 20 can obtain the actual delay duration of audio frames from the source end 10 to the destination end 20 within a set time period. Also, since audio scheduling jitter describes the degree of change in the delay of scheduling audio from the audio buffer to the speaker for playback at the destination end, the destination end 20 can also obtain the callback time interval for the destination end to callback audio frames from the audio buffer 201 within a set time period. The callback time interval refers to the time interval between two adjacent audio frames callbacked by the destination end.

[0051] In the embodiments of the present application, the specific duration of the set time period is not limited. The set time period may be a set parameter adjustment period, such as in the order of seconds (1 second, 2 seconds, 5 seconds, 30 seconds, etc.), minutes (such as 5 minutes, 10 minutes, or half an hour, etc.), hours (such as 1 hour, 2 hours, or 6 hours, etc.), but is not limited thereto.

[0052] Further, the audio quality parameters of the destination end can be determined according to the actual delay duration of the audio frames from the source end 10 to the destination end 20 within the set time period, and the callback time interval for the destination end to callback the audio frames from the audio buffer 201.

[0053] In the embodiments of the present application, the specific implementation manner of determining the audio quality parameters of the destination end according to the above actual delay duration and callback time interval is not limited.

[0054] In some embodiments, the audio transmission jitter parameter can be determined according to the actual delay duration of the audio frames from the source end 10 to the destination end 20 within the set time period. Specifically, in an ideal situation, the actual delay duration of the audio frames from the source end 10 to the destination end 20 is fixed and equal to the preset theoretical delay duration. Among them, the theoretical delay duration is equal to the playing duration of the audio frames at the set playing speed in an ideal situation. The set playing speed can be the original speed of the audio. The original speed of the audio can be the default playing speed, which is generally an empirical value. The degree of fluctuation of the actual delay duration of the audio frames from the source end 10 to the destination end 20 compared to the theoretical delay duration can reflect the audio transmission jitter. Based on this, the audio transmission jitter parameter can be determined according to the actual delay duration of the audio frames from the source end 10 to the destination end 20 and the preset theoretical delay duration within the set time period.

[0055] Optionally, the difference between the actual delay duration of the audio frames from the source end 10 to the destination end 20 and the preset theoretical delay duration within the set time period can be calculated; and the audio transmission jitter parameter can be determined according to the difference between the actual delay duration and the theoretical delay duration of the audio frames from the source end 10 to the destination end 20 within the set time period.

[0056] Specifically, the variance or standard deviation between the actual delay duration and the theoretical delay duration can be calculated according to the difference between the actual delay duration of the audio frames from the source end 10 to the destination end 20 and the theoretical delay duration within the set time period, and used as the audio transmission jitter parameter. Or, the mean value of the difference between the actual delay duration of the audio frames from the source end 10 to the destination end 20 and the theoretical delay duration within the set time period can be calculated and used as the audio transmission jitter parameter.

[0057] In some other embodiments, ideally, the actual delay duration of the audio frames from the source end 10 to the destination end 20 is fixed, and the degree of variation between the actual delay durations can also reflect the audio transmission jitter. Based on this, the average value of the actual delay duration of the audio frames from the source end 10 to the destination end 20 within a set time period can be calculated; and based on the actual delay duration of the audio frames from the source end 10 to the destination end 20 and the average value of the actual delay duration within the set time period, the mean square deviation of the actual delay duration of the audio frames from the source end 10 to the destination end 20 can be calculated as the audio transmission jitter parameter.

[0058] The specific implementation manners of determining the audio transmission jitter parameter according to the actual delay duration of the audio frames from the source end 10 to the destination end 20 shown in the above embodiments are only exemplary descriptions and do not constitute limitations.

[0059] Since the audio scheduling jitter parameter of the destination end is also one of the main parameters of the audio quality parameter of the destination end, therefore, the audio scheduling jitter parameter of the destination end can also be determined according to the callback time interval for the destination end to callback the audio frames from the audio buffer 201 within a set time period.

[0060] Specifically, ideally, the callback time interval of the audio frames is equal to the duration of the audio frames. The duration of the audio frames can be the playing duration of the audio frames at the set playing speed. In the actual process, due to the change in the processor performance of the destination end, the callback time interval of the audio frames may change, and the degree of change in the callback time interval of the audio frames can reflect the audio scheduling jitter of the destination end. Based on this, the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within a set time period can be calculated; and based on the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within the set time period, the audio scheduling jitter parameter can be determined.

[0061] Specifically, the variance or standard deviation between the callback time interval and the duration of the audio frames can be calculated based on the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within a set time period as the audio scheduling jitter parameter. Or, the average value of the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within the set time period can be calculated as the audio scheduling jitter parameter.

[0062] In some other embodiments, ideally, the callback time interval for the destination end to callback audio frames from the audio buffer is fixed, and the degree of variation between the actual callback time intervals can also reflect the audio transmission jitter. Based on this, the average value of the callback time intervals for the destination end to callback audio frames from the audio buffer within a set time period can be calculated; and based on the average value of the callback time intervals for the destination end to callback audio frames from the audio buffer within the set time period, the mean square deviation between the callback time intervals for the destination end to callback audio frames from the audio buffer within the set time period can be calculated as the audio scheduling jitter parameter.

[0063] The calculation method of the audio scheduling jitter parameter shown in the above embodiments is only for illustrative purposes and does not constitute a limitation.

[0064] According to the method shown in the above embodiments, the audio quality parameter of the destination end can be determined. Since the audio stream parameters of the destination end affect the audio quality, the audio stream parameters of the destination end can be adjusted with the audio quality parameter as the feedback condition. Based on this, the destination end 20 can adjust the audio stream parameters of the destination end according to the audio quality parameter of the destination end, realizing the automatic adjustment of the audio stream parameters of the destination end, which helps to improve the efficiency of audio parameter adjustment. On the other hand, adjusting the audio stream parameters of the destination end based on the audio quality parameter helps to ensure the audio quality of the destination end.

[0065] In the embodiments of the present application, the specific implementation content of the audio stream parameters is not limited. In some embodiments, based on the above process of audio playback at the destination end, it can be known that: the larger the capacity of the audio buffer 201, the more audio frames can be stored, and the smaller the time jitter between the destination end obtaining different audio frames by using the callback function, and the smoother the audio playback is perceived by the user at the destination end. Therefore, the data volume of the audio frames stored in the audio buffer 201 has a certain impact on the audio quality of the destination end. However, the more audio frames cached in the audio buffer 201 (i.e., the larger the audio frame data volume), the greater the delay for the destination end 20 to play the audio for the first time. For example, in the cloud desktop or cloud application scenario, the more audio frames cached in the audio buffer of the client, the less affected by network jitter when the client callbacks audio frames from the audio buffer 201, and the higher the audio playback smoothness perceived by the user. The more audio frames cached in the audio buffer 201, the greater the delay of the first frame of the audio data requested by the client for playback.

[0066] Based on the above analysis, the capacity of the audio buffer 201 has a certain impact on the audio playback quality. Based on this, in order to improve the audio playback quality at the destination end, the length of the audio buffer can be used as an audio stream parameter at the destination end. The length of the audio buffer is used to determine the number of audio frames that the audio buffer can store. In the embodiments of the present application, since under the set playback speed, the amount of audio data stored in the audio buffer corresponds one-to-one to the playback duration of the audio. Based on this, the length of the audio buffer can be represented by duration. The capacity of the audio buffer can be equal to the length (duration) of the audio buffer multiplied by the set playback speed.

[0067] Under the set playback speed, the amount of data of an audio frame corresponds to the duration of the audio frame. The greater the duration of the audio frame at the destination end, the less the playback quality of the audio frame is affected by network jitter. Based on this, the length of the audio frame called back at the destination end can also be used as an audio stream parameter at the destination end. Among them, the length of the audio frame can also be represented by duration. The amount of data of the audio frame is equal to the duration of the audio frame multiplied by the set playback speed.

[0068] Based on the above embodiments in which the audio stream parameters at the destination end include the length of the audio buffer at the destination end and the duration of the audio frame, the adjustment of the audio stream parameters at the destination end 20 according to the audio quality parameters can be implemented as: adjusting the length of the audio buffer according to the audio transmission jitter parameter determined according to the above embodiments; and adjusting the length of the audio frame called back at the destination end according to the audio scheduling jitter parameter determined according to the above embodiments.

[0069] In the embodiments of the present application, the specific implementation manners of adjusting the length of the audio buffer according to the audio transmission jitter parameter and adjusting the length of the audio frame called back at the destination end according to the audio scheduling jitter parameter are not limited. Several implementation manners are exemplarily described below.

[0070] In some embodiments, an audio transmission jitter threshold can be preset. The audio transmission jitter threshold is an empirical value obtained according to prior knowledge and is a threshold for determining the stability of audio transmission. Among them, if the audio transmission jitter parameter within a set time period is less than or equal to the audio transmission jitter threshold, it is determined that the audio transmission is stable within the set time period, and the length of the audio buffer can be appropriately reduced, thereby reducing the delay of audio playback at the destination end. If the audio transmission jitter parameter within a set time period is greater than the audio transmission jitter threshold, it is determined that the audio transmission jitters greatly and the audio transmission is unstable, and the length of the audio buffer needs to be increased, thereby reducing the audio transmission jitter.

[0071] Based on this, if the audio transmission jitter parameter within the set time period is less than the set audio transmission jitter threshold, the length of the audio buffer can be adjusted smaller. Optionally, the length of the audio buffer can be adjusted smaller by a set first length based on the original length. The set first length can be flexibly set according to actual adjustment requirements.

[0072] Correspondingly, if the audio transmission jitter parameter within the set time period is less than the set audio transmission jitter threshold, the length of the audio buffer can be adjusted larger.

[0073] In some embodiments, the length of the audio buffer can be adjusted larger by a set second length based on the original length. The set second length can be flexibly set according to actual adjustment requirements. The first length and the second length can be the same or different. Optionally, the first length is less than the second length.

[0074] In other embodiments, the number of audio frames transmitted from the source end to the destination end within the set time period can be obtained; and based on the audio transmission jitter parameter within the above set time period and the number of audio frames transmitted from the source end to the destination end within the set time period, the incremental length of the audio buffer can be determined; and the length of the audio buffer can be increased by the above incremental length based on the original length to obtain the adjusted larger length of the audio buffer.

[0075] For example, based on the audio transmission jitter parameter within the above set time period and the number of audio frames transmitted from the source end to the destination end within the set time period, the average audio transmission jitter of the audio frames within the set time period can be determined as the incremental length of the audio buffer; and the length of the audio buffer can be adjusted larger by the average audio transmission jitter based on the original length to obtain the adjusted larger length of the audio buffer.

[0076] For the embodiment where the variance between the actual delay duration and the theoretical delay duration of the audio frames from the source end to the destination end within the set time period is used as the audio transmission jitter parameter, the average audio transmission jitter of the audio frames within the set time period, that is, the incremental length of the audio buffer, can be expressed as Correspondingly, the adjusted larger length of the audio buffer can be expressed as:

[0077]

[0078] In Equation (1), b1 represents the adjusted larger length of the audio buffer; N represents the audio transmission jitter parameter within the set time period, that is, the variance between the actual delay duration and the theoretical delay duration of the audio frames from the source end to the destination end within the set time period; s represents the number of audio frames transmitted from the source end to the destination end within the set time period. b0 represents the original length of the audio buffer.

[0079] In some other embodiments, an audio scheduling jitter threshold for the destination end may also be preset. The audio scheduling jitter threshold is an empirical value obtained based on prior knowledge and is a threshold for determining the stability of audio scheduling. Among them, if the audio scheduling jitter parameter within a set time period is less than or equal to the audio scheduling jitter threshold, it is determined that the audio scheduling is stable within the set time period, and the length of the audio frames callback at the destination end can be appropriately reduced to reduce the delay of audio playback at the destination end. If the audio scheduling jitter parameter within the set time period is greater than the audio scheduling jitter threshold, it indicates that the audio frames locally callback at the destination end are shorter, resulting in the local processing speed not being able to keep up with the audio frame callback speed. Then, the local audio frame length can be appropriately increased to improve the audio playback stability.

[0080] Based on the above analysis, if the audio scheduling jitter parameter within a set time period is less than the set audio scheduling jitter threshold, the length of the audio frames callback at the destination end can be reduced. Optionally, the length of the audio frames callback at the destination end can be reduced by a set third length on the basis of the original length. The set third length can be flexibly set according to actual adjustment requirements.

[0081] Correspondingly, if the audio scheduling jitter parameter within a set time period is less than the set audio scheduling jitter threshold, the length of the audio frames callback at the destination end can be increased.

[0082] In some embodiments, the length of the audio frames callback at the destination end can be increased by a set fourth length on the basis of the original length. The set fourth length can be flexibly set according to actual adjustment requirements. The third length and the fourth length can be the same or different. Optionally, the third length is less than the fourth length.

[0083] In some other embodiments, the number of audio frames callback at the destination end within a set time period can be obtained; and based on the audio scheduling jitter parameter within the set time period and the number of audio frames callback at the destination end within the set time period, the incremental length of the audio frames callback at the destination end is determined; and the length of the audio frames callback at the destination end is increased by the above incremental length of the audio frames on the basis of the original length to obtain the increased length of the audio frames callback at the target end.

[0084] For example, based on the audio scheduling jitter parameter within the set time period and the number of audio frames callback at the destination end within the set time period, the average audio scheduling jitter of the audio frames within the set time period can be determined as the incremental length of the audio frames callback at the destination end; and the length of the audio buffer, and the length of the audio frames callback at the destination end is increased by the above average audio scheduling jitter on the basis of the original length to obtain the increased length of the audio frames callback at the target end.

[0085] For an embodiment in which the variance between the callback time interval of the destination end and the duration of the audio frame within a set time period is used as the audio scheduling jitter parameter, the average audio callback jitter of the audio frames within the set time period can be expressed as Correspondingly, the increased length of the audio frame callback by the destination end can be expressed as:

[0086]

[0087] In Equation (2), l1 represents the increased length of the audio frame callback by the destination end; L represents the audio scheduling jitter parameter within the set time period, that is, the variance between the callback time interval of the destination end and the duration of the audio frame within the set time period; r represents the number of audio frames callback by the destination end within the set time period. l0 represents the original length of the audio frame callback by the destination end.

[0088] The adjustment methods of the length of the audio buffer at the destination end and the length of the audio frame callback by the destination end shown in the above embodiments are only exemplary descriptions, but do not constitute limitations.

[0089] In addition to the data processing system provided in the above embodiments, the embodiments of the present application also provide an audio parameter adjustment method. The audio parameter adjustment method provided by the embodiments of the present application can be applied to the receiving end of the audio (i.e., the destination end), and can also be applied to other computing devices, such as edge nodes close to the receiving end of the audio in the edge cloud system, etc. The audio parameter adjustment method provided by the embodiments of the present application will be described exemplarily below.

[0090] Figure 3a It is a schematic flowchart of the audio adjustment method provided by the embodiments of the present application. As Figure 3a shown, the audio adjustment method mainly includes:

[0091] 301. Obtain the actual delay duration of the audio frame from the source end to the destination end within a set time period.

[0092] 302. Obtain the callback time interval for the destination end to callback the audio frame from the audio buffer within a set time period.

[0093] 303. Determine the audio quality parameter of the destination end according to the actual delay duration and the callback time interval.

[0094] 304. Adjust the audio stream parameter of the destination end according to the audio quality parameter.

[0095] For the destination end that receives audio data, due to network jitter, the audio frame may not reach the speaker of the destination end on time, resulting in audio stuttering or even disconnection. To reduce the phenomenon of audio stuttering caused by network jitter, etc., the destination end can set an audio buffer. For the description of the audio buffer, reference can be made to the relevant content of the above embodiments.

[0096] Based on the audio buffer, the destination end can obtain the audio stream from the source end, store the audio stream in the audio buffer, and then evenly send the audio stream to the speaker at the destination end for playback through a callback function.

[0097] Based on the above audio buffer, the destination end can obtain the audio frames sent by the source end and store the audio frames sent by the source end in the audio buffer. Further, the audio thread at the destination end can obtain the audio frames to be played from the audio buffer through a callback function; and transmit the audio frames to be played to the speaker at the destination end for playback. This process is the local scheduling process at the destination end.

[0098] In the embodiments of the present application, the audio stream quality mainly reflects the smoothness of video playback at the destination end. The inventors of the present application have found through research that the audio stream quality at the destination end is mainly affected by network jitter and audio scheduling jitter at the destination end itself. Since audio transmission jitter and audio scheduling jitter at the destination end are the main factors affecting the audio stream quality at the destination end, therefore, the audio stream parameters at the destination end can be adjusted according to the audio transmission jitter parameters and audio scheduling jitter parameters.

[0099] Based on the above analysis, since audio transmission jitter describes the degree of change in the delay of audio from the source end to the destination end, therefore, in order to determine the audio quality at the destination end, in step 301, the actual delay duration of audio frames from the source end to the destination end within a set time period can be obtained. Also, since audio scheduling jitter describes the degree of change in the delay of the destination end scheduling audio from the audio buffer to the speaker for playback, therefore, in step 302, the callback time interval for the destination end to callback audio frames from the audio buffer within a set time period can also be obtained.

[0100] Further, in step 303, the audio quality parameters at the destination end can be determined according to the actual delay duration of audio frames from the source end to the destination end within a set time period and the callback time interval for the destination end to callback audio frames from the audio buffer within a set time period.

[0101] In the embodiments of the present application, the specific implementation manner of determining the audio quality parameters at the destination end according to the above actual delay duration and callback time interval is not limited.

[0102] In some embodiments, the audio transmission jitter parameters can be determined according to the actual delay duration of audio frames from the source end to the destination end within a set time period. Specifically, in an ideal situation, the actual delay duration of audio frames from the source end to the destination end is fixed and equal to the preset theoretical delay duration. The degree of fluctuation of the actual delay duration of audio frames from the source end to the destination end compared to the theoretical delay duration can reflect the audio transmission jitter. Based on this, the audio transmission jitter parameters can be determined according to the actual delay duration of audio frames from the source end to the destination end within a set time period and the preset theoretical delay duration.

[0103] Optionally, the difference between the actual delay duration of the audio frames from the source end to the destination end and the preset theoretical delay duration within a set time period can be calculated; and the audio transmission jitter parameter can be determined according to the difference between the actual delay duration of the audio frames from the source end to the destination end and the theoretical delay duration within the set time period.

[0104] Specifically, the variance or standard deviation between the actual delay duration and the theoretical delay duration can be calculated according to the difference between the actual delay duration of the audio frames from the source end to the destination end and the theoretical delay duration within the set time period, and used as the audio transmission jitter parameter. Alternatively, the mean value of the difference between the actual delay duration of the audio frames from the source end to the destination end and the theoretical delay duration within the set time period can be calculated and used as the audio transmission jitter parameter.

[0105] In some other embodiments, ideally, the actual delay duration of the audio frames from the source end to the destination end is fixed, and the degree of variation between the actual delay durations can also reflect the audio transmission jitter situation. Based on this, the mean value of the actual delay duration of the audio frames from the source end to the destination end within a set time period can be calculated; and the mean square deviation between the actual delay durations of the audio frames from the source end to the destination end can be calculated according to the actual delay duration of the audio frames from the source end to the destination end and the mean value of the actual delay duration within the set time period, and used as the audio transmission jitter parameter.

[0106] The specific implementation manners shown in the above embodiments for determining the audio transmission jitter parameter according to the actual delay duration of the audio frames from the source end to the destination end within a set time period are only exemplary descriptions and do not constitute limitations.

[0107] Since the audio scheduling jitter parameter at the destination end is also one of the main parameters of the audio quality parameter at the destination end, therefore, the audio scheduling jitter parameter at the destination end can also be determined according to the callback time interval for the destination end to callback the audio frames from the audio buffer within a set time period.

[0108] Specifically, the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within a set time period can be calculated; and the audio scheduling jitter parameter can be determined according to the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within the set time period.

[0109] Specifically, the variance or standard deviation between the callback time interval and the duration of the audio frames can be calculated according to the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within a set time period, and used as the audio scheduling jitter parameter. Alternatively, the mean value of the difference between the callback time interval for the destination end to callback the audio frames from the audio buffer and the preset duration of the audio frames within the set time period can be calculated and used as the audio scheduling jitter parameter.

[0110] In some other embodiments, ideally, the callback time interval at which the destination end retrieves audio frames from the audio buffer is fixed, and the degree of variation between the actual callback time intervals can also reflect the audio transmission jitter. Based on this, the average value of the callback time intervals at which the destination end retrieves audio frames from the audio buffer within a set time period can be calculated; and based on the average value of the callback time intervals at which the destination end retrieves audio frames from the audio buffer within the set time period, the mean square deviation between the callback time intervals at which the destination end retrieves audio frames from the audio buffer within the set time period is calculated as the audio scheduling jitter parameter.

[0111] The calculation method of the audio scheduling jitter parameter shown in the above embodiments is only for illustrative purposes and does not constitute a limitation.

[0112] According to the method shown in the above embodiments, the audio quality parameter of the destination end can be determined. Since the audio stream parameters of the destination end affect the audio quality, the audio stream parameters of the destination end can be adjusted with the audio quality parameter as the feedback condition. Based on this, in step 304, the audio stream parameters of the destination end can be adjusted according to the audio quality parameter of the destination end, realizing the automatic adjustment of the audio stream parameters of the destination end.

[0113] In this embodiment, the audio quality parameter of the destination end is determined according to the actual delay duration of the audio frames from the source end to the destination end within a set time period and the callback time interval at which the destination end retrieves audio frames from the audio buffer within this time period, realizing the quantification of the audio quality of the destination end. Further, adjusting the audio stream parameters of the destination end according to the audio quality parameter of the destination end realizes the automatic adjustment of the audio stream parameters of the destination end, which helps to improve the efficiency of audio parameter adjustment. On the other hand, adjusting the audio stream parameters of the destination end based on the audio quality parameter provides a certain guarantee for the audio quality of the destination end, which helps to ensure the audio quality of the destination end.

[0114] In the embodiments of the present application, the specific implementation content of the audio stream parameters is not limited. The capacity of the audio buffer has a certain impact on the audio playback quality. Based on this, in order to improve the audio playback quality of the destination end, the length of the audio buffer can be used as the audio stream parameter of the destination end. The length of the audio buffer is used to determine the number of audio frames that the audio buffer can store.

[0115] At a set playback speed, the data volume of the audio frame corresponds to the duration of the audio frame. The greater the duration of the audio frame at the destination end, the less the playback quality of the audio frame is affected by network jitter. Based on this, the duration of the audio frame at the destination end can also be used as the audio stream parameter of the destination end.

[0116] Based on the above embodiments where the audio stream parameters at the destination end include the length of the audio buffer at the destination end and the duration of the audio frame, adjusting the audio stream parameters at the destination end according to the audio quality parameters can be implemented as follows: adjusting the length of the audio buffer according to the audio transmission jitter parameter determined in the above embodiments; and adjusting the length of the audio frame callback at the destination end according to the audio scheduling jitter parameter determined in the above embodiments.

[0117] In the embodiments of the present application, the specific implementation manners of adjusting the length of the audio buffer according to the audio transmission jitter parameter and adjusting the length of the audio frame callback at the destination end according to the audio scheduling jitter parameter are not limited. Several implementation manners are exemplarily described below.

[0118] In some embodiments, an audio transmission jitter threshold may be preset. The audio transmission jitter threshold is an empirical value obtained based on prior knowledge and is a threshold for determining the stability of audio transmission. Among them, if the audio transmission jitter parameter within a set time period is less than or equal to the audio transmission jitter threshold, it is determined that the audio transmission is stable within the set time period, and the length of the audio buffer can be appropriately reduced, thereby reducing the delay of playing audio at the destination end. If the audio transmission jitter parameter within the set time period is greater than the audio transmission jitter threshold, it is determined that the audio transmission jitters greatly and the audio transmission is unstable within the set time period, and the length of the audio buffer needs to be increased, thereby reducing the audio transmission jitter.

[0119] Based on this, if the audio transmission jitter parameter within a set time period is less than the set audio transmission jitter threshold, the length of the audio buffer can be reduced. Optionally, the length of the audio buffer can be reduced by a set first length on the basis of the original length. The set first length can be flexibly set according to actual adjustment requirements.

[0120] Correspondingly, if the audio transmission jitter parameter within a set time period is greater than the set audio transmission jitter threshold, the length of the audio buffer can be increased.

[0121] In some embodiments, the length of the audio buffer can be increased by a set second length on the basis of the original length. The set second length can be flexibly set according to actual adjustment requirements. The first length and the second length may be the same or different. Optionally, the first length is less than the second length.

[0122] In other embodiments, the number of audio frames transmitted from the source end to the destination end within a set time period can be obtained; and according to the audio transmission jitter parameter within the set time period and the number of audio frames transmitted from the source end to the destination end within the set time period, the incremental length of the audio buffer is determined; and the length of the audio buffer is increased by the above incremental length on the basis of the original length to obtain the increased length of the audio buffer.

[0123] For example, the average audio transmission jitter of the audio frames within the set time period can be determined based on the audio transmission jitter parameter within the set time period and the number of audio frames transmitted from the source end to the destination end within the set time period, and used as the incremental length of the audio buffer; and the length of the audio buffer can be increased by the average audio transmission jitter on the basis of the original length to obtain the increased length of the audio buffer.

[0124] In some other embodiments, an audio scheduling jitter threshold of the destination end can also be preset. The audio scheduling jitter threshold is an empirical value obtained based on prior knowledge and is a threshold for determining the stability of audio scheduling. Among them, if the audio scheduling jitter parameter within the set time period is less than or equal to the audio scheduling jitter threshold, it is determined that the audio scheduling is stable within the set time period, and the length of the audio frames called back by the destination end can be appropriately reduced to reduce the delay of audio playback at the destination end. If the audio scheduling jitter parameter within the set time period is greater than the audio scheduling jitter threshold, it means that the audio frames called back locally at the destination end are shorter, resulting in the local processing speed not being able to keep up with the audio frame callback speed, and the length of the local audio frames can be appropriately increased to improve the stability of audio playback.

[0125] Based on the above analysis, if the audio scheduling jitter parameter within the set time period is less than the set audio scheduling jitter threshold, the length of the audio frames called back by the destination end can be reduced. Optionally, the length of the audio frames called back by the destination end can be reduced by a set third length on the basis of the original length. The set third length can be flexibly set according to actual adjustment requirements.

[0126] Correspondingly, if the audio scheduling jitter parameter within the set time period is less than the set audio scheduling jitter threshold, the length of the audio frames called back by the destination end can be increased.

[0127] In some embodiments, the length of the audio frames called back by the destination end can be increased by a set fourth length on the basis of the original length. The set fourth length can be flexibly set according to actual adjustment requirements. The third length and the fourth length are the same or different. Optionally, the third length is less than the fourth length.

[0128] In some other embodiments, the number of audio frames called back by the destination end within the set time period can be obtained; and based on the audio scheduling jitter parameter within the set time period and the number of audio frames called back by the destination end within the set time period, the incremental length of the audio frames called back by the destination end can be determined; and the length of the audio frames called back by the destination end can be increased by the above-mentioned incremental length of the audio frames on the basis of the original length to obtain the increased length of the audio frames called back by the target end.

[0129] For example, based on the audio scheduling jitter parameter within the above-mentioned set time period and the number of audio frames callbacked by the destination end within the set time period, the average audio scheduling jitter of the audio frames within the set time period can be determined as the incremental length of the audio frames callbacked by the destination end; and the length of the audio buffer, and the length of the audio frames callbacked by the destination end are increased by the above-mentioned average audio scheduling jitter on the basis of the original length to obtain the increased length of the audio frames callbacked by the target end.

[0130] The audio parameter adjustment method provided by the embodiments of the present application is applicable to application scenarios with high requirements for audio real-time performance. For example, it can be applicable to cloud desktop or cloud application scenarios. In cloud desktop or cloud application scenarios, the above-mentioned source end is the server of the cloud desktop or the server of the cloud application; the client is the cloud desktop client or the cloud application client. The server can store the audio data of the cloud-side application and send the audio data to the client in response to the request of the client. In an audio and video call scenario, the source end can collect the audio data of the source end user through a pickup and send the audio data to other terminals (i.e., the destination end) in this call. Taking the cloud desktop or cloud application scenario as an example below, an exemplary description of the audio parameter adjustment method provided by the embodiments of the present application is given.

[0131] Figure 3b It is a schematic flowchart of another audio parameter adjustment method provided by the embodiments of the present application. This method can be applicable to the client of cloud desktop or cloud application, and can also be applicable to other computing devices, such as edge nodes on the client side in an edge cloud system, etc. As Figure 3b shown, this method mainly includes:

[0132] 31. Obtain the actual delay duration of the audio frames from the server to the client within the set time period; the server and the client are respectively the cloud desktop server and the cloud desktop client, or the cloud application server and the cloud desktop client.

[0133] 32. Obtain the callback time interval for the client to callback audio frames from the audio buffer within the set time period.

[0134] 33. Determine the audio quality parameter of the client according to the actual delay duration and the callback time interval.

[0135] 34. Adjust the audio stream parameter of the client according to the audio quality parameter.

[0136] For the client receiving audio data in cloud desktop or cloud application scenarios, due to network jitter, audio frames may not reach the client's speaker on time, resulting in audio stuttering or even disconnection. To reduce the phenomenon of audio stuttering caused by network jitter, etc., the client can set an audio buffer. For the description of the audio buffer, reference can be made to the relevant content of the above embodiments.

[0137] Since the audio transmission jitter describes the degree of variation in the latency of audio from the server side to the client side, in order to determine the audio quality of the client, in step 31, the actual latency duration of audio frames from the cloud desktop server to the cloud desktop client within a set time period can be obtained; alternatively, the actual latency duration of audio frames from the cloud application server to the cloud application client within a set time period can be obtained. Also, since the audio scheduling jitter describes the degree of variation in the latency of the client scheduling audio from the audio buffer to the speaker for playback, in step 32, the callback time interval for the client to callback audio frames from the audio buffer within a set time period can also be obtained.

[0138] Further, in step 33, the audio quality parameter of the client can be determined based on the actual latency duration of audio frames from the server side to the client side within a set time period and the callback time interval for the client to callback audio frames from the audio buffer within a set time period.

[0139] In the embodiment of the present application, for the specific implementation manner of determining the audio quality parameter of the client according to the above actual latency duration and callback time interval, reference can be made to the relevant content of determining the audio quality parameter of the destination end according to the actual latency duration and callback time interval above, which will not be elaborated here.

[0140] Since the audio stream parameters of the client affect the audio quality, the audio stream parameters of the client can be adjusted with the audio quality parameter as the feedback condition. Based on this, in step 34, the audio stream parameters of the client can be adjusted according to the audio quality parameter of the client, realizing the automatic adjustment of the audio stream parameters of the client.

[0141] In this embodiment, the audio quality parameter of the client is determined based on the actual latency duration of audio frames from the server side to the client side within a set time period and the callback time interval for the client to callback audio frames from the audio buffer within this time period, realizing the quantification of the audio quality of the client. Further, the audio stream parameters of the client are adjusted according to the audio quality parameter of the client, realizing the automatic adjustment of the audio stream parameters of the client, which helps to improve the efficiency of audio parameter adjustment. On the other hand, adjusting the audio stream parameters of the client based on the audio quality parameter provides a certain guarantee for the audio quality of the client, which helps to ensure the audio quality of the client.

[0142] In the embodiment of the present application, the specific implementation content of the audio stream parameters is not limited. The capacity of the audio buffer has a certain impact on the audio playback quality. Based on this, in order to improve the audio playback quality of the client, the length of the audio buffer can be used as the audio stream parameter of the client. The length of the audio buffer is used to determine the number of audio frames that the audio buffer can store.

[0143] At the set playback speed, the data volume of an audio frame corresponds to the duration of the audio frame. The greater the duration of the audio frame on the client side, the less the playback quality of the audio frame is affected by network jitter. Based on this, the duration of the audio frame on the client side can also be used as an audio stream parameter of the client.

[0144] Based on the above embodiments in which the audio stream parameters of the client include the length of the audio buffer of the client and the duration of the audio frame, according to the audio quality parameters, adjusting the audio stream parameters of the client can be implemented as: adjusting the length of the audio buffer according to the audio transmission jitter parameter determined in the above embodiments; and adjusting the length of the audio frame called back by the client according to the audio scheduling jitter parameter determined in the above embodiments.

[0145] In the embodiments of the present application, for the specific implementation manners of adjusting the length of the audio buffer according to the audio transmission jitter parameter and adjusting the length of the audio frame called back by the client according to the audio scheduling jitter parameter, reference may be made to the relevant content of the above embodiments, which will not be elaborated here.

[0146] It should be noted that the execution subject of each step of the method provided in the above embodiments can be the same device, or the method can also be executed by different devices as the execution subject. For example, the execution subjects of steps 301 and 302 can be device A; for another example, the execution subject of step 301 can be device A, and the execution subject of step 302 can be device B; and so on.

[0147] In addition, in some of the processes described in the above embodiments and the accompanying drawings, a plurality of operations appear in a specific order, but it should be clearly understood that these operations can be executed not in the order in which they appear in this article or in parallel. The operation numbers such as 301 and 302 are only used to distinguish different operations, and the numbers themselves do not represent any execution order. In addition, these processes can include more or fewer operations, and these operations can be executed in sequence or in parallel.

[0148] Correspondingly, the embodiments of the present application further provide a computer-readable storage medium storing computer instructions, which when executed by one or more processors, cause the one or more processors to execute the steps in the above various audio parameter adjustment methods.

[0149] Figure 4 It is a schematic structural diagram of a computing device provided by the embodiments of the present application. As Figure 4 shown, the computing device mainly includes: a memory 40a and a processor 40b. The memory 40a is used to store computer programs.

[0150] The processor 40b is coupled to the memory 40a and is configured to execute a computer program for: obtaining the actual delay duration of audio frames from a source end to a destination end within a set time period; obtaining the callback time interval for the destination end to callback audio frames from an audio buffer; determining an audio quality parameter of the destination end according to the actual delay duration and the callback time interval; and adjusting an audio stream parameter of the destination end according to the audio quality parameter.

[0151] In some embodiments, the audio quality parameter includes: an audio transmission jitter parameter and an audio scheduling jitter parameter of the destination end. Accordingly, when determining the audio quality parameter of the destination end according to the actual delay duration and the callback time interval, the processor 40b is specifically configured to: determine the audio transmission jitter parameter according to the actual delay duration and a preset theoretical delay duration; and determine the audio scheduling jitter parameter of the destination end according to the callback time interval and the duration of an audio frame.

[0152] Further, when determining the audio transmission jitter parameter according to the actual delay duration and the preset theoretical delay duration, the processor 40b is specifically configured to: calculate the difference between the actual delay duration and the theoretical delay duration; and determine the audio transmission jitter parameter according to the difference between the actual delay duration and the theoretical delay duration.

[0153] Optionally, when determining the audio transmission jitter parameter according to the difference between the actual delay duration and the theoretical delay duration, the processor 40b is specifically configured to: calculate the variance or standard deviation between the actual delay duration and the theoretical delay duration as the audio transmission jitter parameter according to the difference between the actual delay duration and the theoretical delay duration; or calculate the mean value of the difference between the actual delay duration and the theoretical delay duration as the audio transmission jitter parameter.

[0154] In some other embodiments, when determining the audio scheduling jitter parameter of the destination end according to the callback time interval and the duration of an audio frame, the processor 40b is specifically configured to: calculate the difference between the callback time interval and the duration of an audio frame; and determine the audio scheduling jitter parameter according to the difference between the callback time interval and the duration of an audio frame.

[0155] Further, when determining the audio scheduling jitter parameter according to the difference between the callback time interval and the duration of an audio frame, the processor 40b is specifically configured to: calculate the variance or standard deviation between the callback time interval and the duration of an audio frame as the audio transmission jitter parameter according to the difference between the callback time interval and the duration of an audio frame; or calculate the mean value of the difference between the callback time interval and the duration of an audio frame as the audio transmission jitter parameter.

[0156] In some embodiments of the present application, the audio stream parameters include: the length of the audio buffer and the length of the audio frames at the local end of the destination. Accordingly, when adjusting the audio stream parameters at the destination according to the audio quality parameters, the processor 40b is specifically configured to: adjust the length of the audio buffer according to the audio transmission jitter parameter; and adjust the length of the audio frames called back at the destination according to the audio scheduling jitter parameter.

[0157] Further, when adjusting the length of the audio buffer according to the audio transmission jitter parameter, the processor 40b is specifically configured to: if the audio transmission jitter parameter is less than the set audio transmission jitter threshold, reduce the length of the audio buffer; if the audio transmission jitter parameter is greater than the audio transmission jitter threshold, increase the length of the audio buffer.

[0158] Optionally, when reducing the length of the audio buffer, the processor 40b is specifically configured to: reduce the length of the audio buffer by a set first length on the basis of the original length.

[0159] And / or, when increasing the length of the audio buffer, the processor 40b is specifically configured to: obtain the first number of audio frames transmitted from the source end to the destination end within a set time period; determine the first incremental length corresponding to the audio buffer according to the audio transmission jitter parameter and the first number; increase the length of the audio buffer by the first incremental length on the basis of the original length.

[0160] Further, when increasing the length of the audio buffer according to the audio transmission jitter parameter and the first number, the processor 40b is specifically configured to: calculate the average audio transmission jitter corresponding to each audio frame within a set time period according to the audio transmission jitter parameter and the first number, and use it as the first incremental length.

[0161] Optionally, when adjusting the length of the audio frames called back at the destination according to the audio scheduling jitter parameter, the processor 40b is specifically configured to: if the audio scheduling jitter parameter is less than the set audio scheduling jitter threshold, reduce the length of the audio frames called back at the destination; if the audio transmission jitter parameter is greater than the audio scheduling jitter threshold, increase the length of the audio frames called back at the destination.

[0162] Optionally, when reducing the length of the audio frames called back at the destination, the processor 40b is specifically configured to: reduce the length of the audio frames called back at the destination by a set second length on the basis of the original length.

[0163] And / or, when increasing the length of the audio frames called back at the destination, the processor 40b is specifically configured to: obtain the second number of audio frames called back from the destination within a set time period; determine the second incremental length corresponding to the audio frames called back at the destination according to the audio scheduling jitter parameter and the second number; increase the length of the audio frames called back at the destination by the second incremental length on the basis of the original length.

[0164] Optionally, when the processor 40b increases the length of the audio frame called back at the destination end according to the audio scheduling jitter parameter and the second quantity, it is specifically configured to: calculate the average audio callback jitter corresponding to each audio frame within a set time period according to the audio scheduling jitter parameter and the first quantity, and use it as the second incremental length.

[0165] In some embodiments, the computing device is implemented as the above-mentioned destination end. Correspondingly, the computing device may further include: a communication component 40c and an audio component 40d. The processor 40b is further configured to: obtain the audio frame sent by the source end through the communication component 40d; store the audio frame sent by the source end in the audio buffer; use the audio thread at the destination end to obtain the audio frame to be played from the audio buffer through a callback function; and transmit the audio frame to be played to the speaker in the audio component 40d for playing.

[0166] In some embodiments of the present application, the processor 40b is further configured to: obtain the actual delay duration of the audio frame from the server end to the client end within a set time period; the server end and the client end are a cloud desktop server end and a cloud desktop client end, or a cloud application server end and a cloud application client end; obtain the callback time interval for the client to call back the audio frame from the audio buffer within a set time period; determine the audio quality parameter of the client according to the actual delay duration and the callback time interval; and adjust the audio stream parameter of the client according to the audio quality parameter. For the specific implementation manners of the operation steps in this embodiment, reference may be made to the relevant content of the above embodiments, which will not be elaborated here.

[0167] In some alternative embodiments, as Figure 4 shown, the computing device may further include: a power supply component 40e, a display component 40f and other components. Figure 4 Only some components are schematically shown, which does not mean that the computing device must include Figure 4 all the components shown, nor does it mean that the computing device can only include Figure 4 the components shown.

[0168] The computing device provided in this embodiment can determine the audio quality parameter of the destination end according to the actual delay duration of the audio frame from the source end to the destination end within a set time period and the callback time interval for the destination end to call back the audio frame from the audio buffer within this time period, realizing the quantification of the audio quality at the destination end. Further, adjusting the audio stream parameter of the destination end according to the audio quality parameter of the destination end realizes the automatic adjustment of the audio stream parameter of the destination end, which helps to improve the audio parameter adjustment efficiency. On the other hand, adjusting the audio stream parameter of the destination end based on the audio quality parameter ensures a certain degree of audio quality at the destination end, which helps to ensure the audio quality at the destination end.

[0169] Figure 5The structural schematic diagram of the audio parameter adjustment device provided by the embodiment of the present application is as follows. As Figure 5 shown, the audio parameter adjustment device includes: an acquisition module 50a, a determination module 50b, and an adjustment module 50c.

[0170] Among them, the acquisition module 50a is used to acquire the actual delay duration of the audio frame from the source end to the destination end within a set time period; and, acquire the callback time interval for the destination end to callback the audio frame from the audio buffer within the set time period.

[0171] The determination module 50b is used to determine the audio quality parameters of the destination end according to the actual delay duration and the callback time interval.

[0172] The adjustment module 50c is used to adjust the audio stream parameters of the destination end according to the audio quality parameters.

[0173] In some embodiments, the audio quality parameters include: the audio transmission jitter parameter and the audio scheduling jitter parameter of the destination end. Correspondingly, when the determination module 50b determines the audio quality parameters of the destination end according to the actual delay duration and the callback time interval, it specifically is used for: determining the audio transmission jitter parameter according to the difference between the actual delay duration and the preset theoretical delay duration; determining the audio scheduling jitter parameter of the destination end according to the difference between the callback time interval and the duration of the audio frame.

[0174] Optionally, when the determination module 50b determines the audio transmission jitter parameter according to the difference between the actual delay duration and the theoretical delay duration, it specifically is used for: calculating the variance or standard deviation between the actual delay duration and the theoretical delay duration according to the difference between the actual delay duration and the theoretical delay duration, as the audio transmission jitter parameter; or calculating the mean value of the difference between the actual delay duration and the theoretical delay duration, as the audio transmission jitter parameter.

[0175] Optionally, when the determination module 50b determines the audio scheduling jitter parameter according to the difference between the callback time interval and the duration of the audio frame, it specifically is used for: calculating the variance or standard deviation between the callback time interval and the duration of the audio frame according to the difference between the callback time interval and the duration of the audio frame, as the audio transmission jitter parameter; or calculating the mean value of the difference between the callback time interval and the duration of the audio frame, as the audio transmission jitter parameter.

[0176] In some other embodiments, the audio stream parameters include: the length of the audio buffer and the length of the audio frame at the destination end locally. Correspondingly, when the adjustment module 50c adjusts the audio stream parameters of the destination end according to the audio quality parameters, it specifically is used for: adjusting the length of the audio buffer according to the audio transmission jitter parameter; adjusting the length of the audio frame callbacked by the destination end according to the audio scheduling jitter parameter.

[0177] Further, when adjusting the length of the audio buffer according to the audio transmission jitter parameter, the adjustment module 50c is specifically configured to: if the audio transmission jitter parameter is less than the set audio transmission jitter threshold, reduce the length of the audio buffer; if the audio transmission jitter parameter is greater than the audio transmission jitter threshold, increase the length of the audio buffer.

[0178] Optionally, when reducing the length of the audio buffer, the adjustment module 50c is specifically configured to: reduce the length of the audio buffer by a set first length based on the original length.

[0179] When increasing the length of the audio buffer, the adjustment module 50c is specifically configured to: obtain a first quantity of audio frames transmitted from the source end to the destination end within a set time period; determine a first incremental length corresponding to the audio buffer according to the audio transmission jitter parameter and the first quantity; and increase the length of the audio buffer by the first incremental length based on the original length.

[0180] Optionally, when adjusting the length of the audio frames called back at the destination end according to the audio scheduling jitter parameter, the adjustment module 50c is specifically configured to: if the audio scheduling jitter parameter is less than the set audio scheduling jitter threshold, reduce the length of the audio frames called back at the destination end; if the audio transmission jitter parameter is greater than the audio scheduling jitter threshold, increase the length of the audio frames called back at the destination end.

[0181] Further, when reducing the length of the audio frames called back at the destination end, the adjustment module 50c is specifically configured to: reduce the length of the audio frames called back at the destination end by a set second length based on the original length.

[0182] When increasing the length of the audio frames called back at the destination end, the adjustment module 50c includes: obtaining a second quantity of audio frames called back at the destination end within a set time period; determining a second incremental length corresponding to the audio frames called back at the destination end according to the audio scheduling jitter parameter and the second quantity; and increasing the length of the audio frames called back at the destination end by the second incremental length based on the original length.

[0183] In some embodiments of the present application, the acquisition module 50a is configured to acquire the actual delay duration of audio frames from the server end to the client end within a set time period, where the server end and the client end are a cloud desktop server end and a cloud desktop client end, or a cloud application server end and a cloud application client end; and, acquire the callback time interval for the client to call back audio frames from the audio buffer within a set time period.

[0184] The determination module 50b is configured to determine the audio quality parameter of the client according to the actual delay duration and the callback time interval.

[0185] The adjustment module 50c is configured to adjust the audio stream parameter of the client according to the audio quality parameter.

[0186] Regarding the specific implementation manner of the acquisition module 50a to acquire the actual latency duration of audio frames from the server of the cloud desktop or cloud application to the client of the cloud desktop or cloud application within a set time period, and to acquire the callback time interval for the client to callback audio frames from the audio buffer, the specific implementation manner of the determination module 50b to determine the audio quality parameter of the client according to the actual latency duration and the callback time interval, and the specific implementation manner of the adjustment module 50c to adjust the audio stream parameter of the client according to the audio quality parameter, reference may be made to the relevant content of the above embodiments, which will not be elaborated here.

[0187] The audio parameter adjustment device provided in this embodiment can determine the audio quality parameter of the destination end according to the actual latency duration of audio frames from the source end to the destination end within a set time period and the callback time interval for the destination end to callback audio frames from the audio buffer within this time period, realizing the quantification of the audio quality of the destination end. Further, according to the audio quality parameter of the destination end, adjusting the audio stream parameter of the destination end realizes the automatic adjustment of the audio stream parameter of the destination end, which helps to improve the audio parameter adjustment efficiency. On the other hand, adjusting the audio stream parameter of the destination end based on the audio quality parameter ensures a certain level of the audio quality of the destination end, which helps to guarantee the audio quality of the destination end.

[0188] In the embodiment of the present application, the memory is used to store computer programs and can be configured to store various other data to support the operations on the device where it is located. Among them, the processor can execute the computer programs stored in the memory to implement the corresponding control logic. The memory can be implemented by any type of volatile or non-volatile storage device or a combination thereof, such as Static Random-Access Memory (SRAM), Electrically Erasable Programmable Read Only Memory (EEPROM), Electrical Programmable Read Only Memory (EPROM), Programmable Read Only Memory (PROM), Read Only Memory (ROM), magnetic memory, flash memory, magnetic disk or optical disk.

[0189] In the embodiments of the present application, the processor may be any hardware processing device capable of executing the above method logic. Optionally, the processor may be a Central Processing Unit (CPU), a Graphics Processing Unit (GPU), or a Microcontroller Unit (MCU); it may also be a programmable device such as a Field-Programmable Gate Array (FPGA), a Programmable Array Logic (PAL), a General Array Logic (GAL), or a Complex Programmable Logic Device (CPLD); or an Application Specific Integrated Circuit (ASIC) chip; or an Advanced Reduced Instruction Set Compute (RISC) processor (Advanced RISC Machines, ARM) or a System on Chip (SoC), etc., but not limited thereto.

[0190] In the embodiments of the present application, the communication component is configured to facilitate communication between the device where it is located and other devices in a wired or wireless manner. The device where the communication component is located can access a wireless network based on communication standards, such as Wireless Fidelity (WiFi), 2G or 3G, 4G, 5G, or a combination thereof. In an exemplary embodiment, the communication component receives a broadcast signal or broadcast-related information from an external broadcast management system via a broadcast channel. In an exemplary embodiment, the communication component can also be implemented based on Near Field Communication (NFC) technology, Radio Frequency Identification (RFID) technology, Infrared Data Association (IrDA) technology, Ultra Wide Band (UWB) technology, Bluetooth (BT) technology, or other technologies.

[0191] In an embodiment of the present application, the display component may include a Liquid Crystal Display (LCD) and a Touch Panel (TP). If the display component includes a touch panel, the display component can be implemented as a touch screen to receive input signals from a user. The touch panel includes one or more touch sensors to sense touches, swipes, and gestures on the touch panel. The touch sensors can sense not only the boundaries of touch or swipe actions but also detect the duration and pressure associated with the touch or swipe operations.

[0192] In an embodiment of the present application, the power supply component is configured to provide power to various components of the device where it is located. The power supply component may include a power management system, one or more power supplies, and other components associated with generating, managing, and distributing power for the device where the power supply component is located.

[0193] In an embodiment of the present application, the audio component can be configured to output and / or input audio signals. For example, the audio component includes a microphone (MIC). When the device where the audio component is located is in an operating mode, such as a call mode, a recording mode, and a voice recognition mode, the microphone is configured to receive external audio signals. The received audio signals can be further stored in the memory or sent via the communication component. In some embodiments, the audio component further includes a speaker for outputting audio signals. For example, for a device with a language interaction function, voice interaction with the user can be achieved through the audio component.

[0194] It should be noted that the user information (including but not limited to user device information, user personal information, etc.) and data (including but not limited to data for analysis, stored data, displayed data, etc.) involved in the present application are all information and data that have been authorized by the user or fully authorized by all parties. And the collection, use, and processing of relevant data need to comply with the relevant laws, regulations, and standards of relevant countries and regions, and corresponding operation entrances are provided for users to choose to authorize or refuse.

[0195] It should also be noted that the descriptions such as "first" and "second" in this article are used to distinguish different messages, devices, modules, etc., and do not represent a sequence, nor do they limit that "first" and "second" are of different types.

[0196] Those skilled in the art should understand that the embodiments of the present application can be provided as a method, a system, or a computer program product. Therefore, the present application can take the form of a complete hardware embodiment, a complete software embodiment, or an embodiment combining software and hardware aspects. Moreover, the present application can take the form of a computer program product implemented on one or more computer-usable storage media (including but not limited to disk memory, Compact Disc Read-Only Memory (CD-ROM), optical memory, etc.) that contain computer-usable program code.

[0197] The present application is described with reference to the flowcharts and / or block diagrams of methods, devices (or systems), and computer program products according to the embodiments of the present application. It should be understood that each flow and / or block in the flowchart and / or block diagram, as well as the combination of flows and / or blocks in the flowchart and / or block diagram, can be implemented by computer program instructions. These computer program instructions can be provided to the processor of a general-purpose computer, a special-purpose computer, an embedded processor, or other programmable data processing devices to generate a machine, such that the instructions executed by the processor of the computer or other programmable data processing devices generate a device for implementing the functions specified in Figure 1 one flow or multiple flows and / or blocks Figure 1 one block or multiple blocks.

[0198] These computer program instructions can also be stored in a computer-readable memory that can direct a computer or other programmable data processing device to work in a specific manner, such that the instructions stored in the computer-readable memory generate a manufactured article including an instruction device that implements the functions specified in Figure 1 one flow or multiple flows and / or blocks Figure 1 one block or multiple blocks.

[0199] These computer program instructions can also be loaded onto a computer or other programmable data processing device, such that a series of operation steps are executed on the computer or other programmable device to generate a computer-implemented process, and thus the instructions executed on the computer or other programmable device provide steps for implementing the functions specified in Figure 1 one flow or multiple flows and / or blocks Figure 1 one block or multiple blocks.

[0200] In a typical configuration, a computing device includes one or more processors (such as a CPU, etc.), an input / output interface, a network interface, and a memory.

[0201] Memory may include non-permanent memory in the form of computer-readable media, such as random access memory (RAM) and / or non-volatile memory, such as read only memory (ROM) or flash RAM. Memory is an example of computer-readable media.

[0202] The storage medium of a computer is a readable storage medium, also known as a readable medium. Readable storage media include permanent and non-permanent, removable and non-removable media and information storage can be implemented by any method or technology. The information can be computer-readable instructions, data structures, program modules or other data. Examples of computer storage media include, but are not limited to, phase change memory (PRAM), static random access memory (SRAM), dynamic random access memory (DRAM), other types of random access memory (RAM), read only memory (ROM), electrically erasable programmable read only memory (EEPROM), flash memory or other memory technologies, CD-ROM, digital video disc (DVD) or other optical storage, magnetic cassette tapes, disk storage or other magnetic storage devices, or any other non-transmission media that can be used to store information that can be accessed by a computing device. As defined herein, computer-readable media does not include transitory media such as modulated data signals and carrier waves.

[0203] It should also be noted that the term "comprising", "including" or any other variation thereof is intended to cover non-exclusive inclusion, such that a process, method, article or device comprising a series of elements includes not only those elements but also other elements not expressly listed, or elements inherent to such process, method, article or device. Without further limitation, an element defined by the statement "comprising an..." does not exclude the presence of additional identical elements in the process, method, article or device comprising the above elements.

[0204] The above content is only an embodiment of the present application and is not intended to limit the present application. For those skilled in the art, various changes and modifications can be made to the present application. Any modification, equivalent replacement, improvement, etc. made within the spirit and principle of the present application shall be included within the scope of the claims of the present application.

Claims

1. An audio parameter adjustment method, characterized in that, Including: Obtain the actual delay duration of the audio frames from the source end to the destination end within a set time period; Obtain the callback time interval for the destination end to callback audio frames from the audio buffer within the set time period; the callback time interval refers to the time interval between two adjacent audio frames callbacked by the destination end; Determine the audio quality parameters of the destination end according to the actual delay duration and the callback time interval; Adjust the audio stream parameters of the destination end according to the audio quality parameters; The audio stream parameters include: the length of the audio buffer and the length of the audio frames on the local side of the destination end; the audio quality parameters include: the audio transmission jitter parameter and the audio scheduling jitter parameter of the destination end; Wherein, the adjusting the audio stream parameters of the destination end according to the audio quality parameters includes: Adjust the length of the audio buffer according to the audio transmission jitter parameter; Adjust the length of the audio frames callbacked by the destination end according to the audio scheduling jitter parameter.

2. The method according to claim 1, wherein The determining the audio quality parameters of the destination end according to the actual delay duration and the callback time interval includes: Determine the audio transmission jitter parameter according to the difference between the actual delay duration and the preset theoretical delay duration; Determine the audio scheduling jitter parameter of the destination end according to the difference between the callback time interval and the duration of the audio frames.

3. The method according to claim 2, wherein The determining the audio transmission jitter parameter according to the difference between the actual delay duration and the theoretical delay duration includes: Calculate the variance or standard deviation between the actual delay duration and the theoretical delay duration according to the difference between the actual delay duration and the theoretical delay duration, and use it as the audio transmission jitter parameter; Or, Calculate the mean value of the difference between the actual delay duration and the theoretical delay duration, and use it as the audio transmission jitter parameter.

4. The method according to claim 2, wherein The determining the audio scheduling jitter parameter according to the difference between the callback time interval and the duration of the audio frames includes: Calculate the variance or standard deviation between the callback time interval and the duration of the audio frames according to the difference between the callback time interval and the duration of the audio frames, and use it as the audio transmission jitter parameter; Or, Calculate the mean value of the difference between the callback time interval and the duration of the audio frames, and use it as the audio transmission jitter parameter.

5. The method according to claim 1, wherein The adjusting the length of the audio buffer according to the audio transmission jitter parameter includes: If the audio transmission jitter parameter is less than the set audio transmission jitter threshold, reduce the length of the audio buffer; If the audio transmission jitter parameter is greater than the audio transmission jitter threshold, increase the length of the audio buffer.

6. The method according to claim 5, characterized in that The reducing the length of the audio buffer includes: Reduce the length of the audio buffer by a set first length on the basis of the original length; And / or, The increasing the length of the audio buffer includes: Obtain the first quantity of the audio frames transmitted from the source end to the destination end within the set time period; Determine the first incremental length corresponding to the audio buffer according to the audio transmission jitter parameter and the first quantity; Increase the length of the audio buffer by the first increment length based on the original length.

7. The method according to claim 1, wherein The adjusting the length of the audio frames called back at the destination end according to the audio scheduling jitter parameter includes: If the audio scheduling jitter parameter is less than the set audio scheduling jitter threshold, reduce the length of the audio frames called back at the destination end; If the audio transmission jitter parameter is greater than the audio scheduling jitter threshold, increase the length of the audio frames called back at the destination end.

8. The method according to claim 7, wherein The reducing the length of the audio frames called back at the destination end includes: Reduce the length of the audio frames called back at the destination end by a set second length based on the original length; and / or The increasing the length of the audio frames called back at the destination end includes: Obtain the second quantity of the audio frames called back at the destination end within the set time period; Determine the second increment length corresponding to the audio frames called back at the destination end according to the audio scheduling jitter parameter and the second quantity; Increase the length of the audio frames called back at the destination end by the second increment length based on the original length.

9. An audio parameter adjustment method, characterized in that, including: Obtain the actual delay duration of the audio frames from the server end to the client end within the set time period; The server end and the client end are the cloud desktop server end and the cloud desktop client end, or the cloud application server end and the cloud desktop client end; Obtain the callback time interval for the client to call back audio frames from the audio buffer within the set time period; the callback time interval refers to the time interval between two adjacent audio frames called back at the destination end; Determine the audio quality parameter of the client according to the actual delay duration and the callback time interval; Adjust the audio stream parameter of the client according to the audio quality parameter; The audio stream parameter includes: the length of the audio buffer and the length of the audio frames at the local end of the destination end; the audio quality parameter includes: the audio transmission jitter parameter and the audio scheduling jitter parameter of the destination end; Wherein, the adjusting the audio stream parameter of the destination end according to the audio quality parameter includes: Adjust the length of the audio buffer according to the audio transmission jitter parameter; Adjust the length of the audio frames called back at the destination end according to the audio scheduling jitter parameter.

10. An audio parameter adjustment device, characterized in that, including: An obtaining module, configured to obtain the actual delay duration of the audio frames from the source end to the destination end within the set time period; and, obtain the callback time interval for the destination end to call back audio frames from the audio buffer within the set time period; the callback time interval refers to the time interval between two adjacent audio frames called back at the destination end; A determining module, configured to determine the audio quality parameter of the destination end according to the actual delay duration and the callback time interval; An adjusting module, configured to adjust the audio stream parameter of the destination end according to the audio quality parameter; The audio stream parameter includes: the length of the audio buffer and the length of the audio frames at the local end of the destination end; the audio quality parameter includes: the audio transmission jitter parameter and the audio scheduling jitter parameter of the destination end; Wherein, the adjusting the audio stream parameter of the destination end according to the audio quality parameter includes: Adjust the length of the audio buffer according to the audio transmission jitter parameter; Adjust the length of the audio frames called back at the destination end according to the audio scheduling jitter parameter.

11. An audio parameter adjustment device, characterized in that, Including: An acquisition module, configured to acquire the actual delay duration of the audio frames from the server to the client within a set time period; And acquire the callback time interval for the client to call back audio frames from the audio buffer within the set time period; The server and the client are a cloud desktop server and a cloud desktop client, or a cloud application server and a cloud desktop client; the callback time interval refers to the time interval between two adjacent audio frames called back at the destination end; A determination module, configured to determine the audio quality parameter of the client according to the actual delay duration and the callback time interval; An adjustment module, configured to adjust the audio stream parameter of the client according to the audio quality parameter; The audio stream parameter includes: the length of the audio buffer and the length of the audio frames at the local end of the destination end; the audio quality parameter includes: an audio transmission jitter parameter and the audio scheduling jitter parameter of the destination end; Wherein, the adjusting the audio stream parameter of the destination end according to the audio quality parameter includes: Adjust the length of the audio buffer according to the audio transmission jitter parameter; Adjust the length of the audio frames called back at the destination end according to the audio scheduling jitter parameter.

12. A computing device, characterized in that, Including: A memory and a processor; wherein, the memory is used to store a computer program; The processor is coupled to the memory and is configured to execute the computer program to perform the steps in the method according to any one of claims 1-9.

13. A computer-readable storage medium storing computer instructions, characterized in that, When the computer instructions are executed by one or more processors, the one or more processors are caused to perform the steps in the method according to any one of claims 1-9.

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