CN117938822A - WASM-based real-time voice communication method, terminal and camera - Google Patents
WASM-based real-time voice communication method, terminal and camera Download PDFInfo
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Abstract
The invention belongs to the technical field of voice communication, and discloses a WASM-based real-time voice communication method, a terminal and a camera, wherein the method comprises the following steps: collecting original audio data; encoding the original audio data based on a preset function instance to obtain an audio stream in a preset format; generating a receiving control request of an audio stream with a preset format; the method comprises the steps of sending an audio stream with a preset format to a streaming media server and sending a receiving control request to a control platform, wherein the control platform is used for generating an audio stream pulling instruction according to the receiving control request and sending the audio stream pulling instruction to a camera based on SIP signaling. The communication method can complete the processing of the audio data more quickly, and has lower audio transmission delay; and can run on any browser, and has good portability and compatibility.
Description
Technical Field
The invention belongs to the technical field of voice communication, and particularly relates to a WASM-based real-time voice communication method, a terminal and a camera.
Background
Real-time speech technology is a technology that is capable of immediately converting human speech into text or for other processing. It can be widely used in various fields such as speech recognition, speech translation, speech synthesis, etc. The development of the real-time voice technology has important significance in the aspects of information acquisition promotion, communication efficiency improvement, intelligent interaction realization and the like.
IPC network cameras, i.e., network video monitoring cameras, are a type of monitoring device that performs video transmission over a network, and these cameras use the internet protocol for capturing, encoding, transmitting, and storing video data. The existing IPC network camera supports real-time two-way voice call, can receive voice messages of clients, and can be applied to the fields of home monitoring, remote office, video conference, security monitoring and the like.
At present, a common implementation method of real-time voice between an IPC network camera and a client is as follows:
The method comprises the following steps: the realization method based on the server and the client model comprises the following steps: in this method, voice data is collected from one end and then transmitted to the other end for playing through a network, and this method requires real-time communication between a server and a client, and may use TCP or UDP protocols;
And two,: the implementation method based on WebRTC comprises the following steps: webRTC (Web real-time communication) is a real-time communication technology, which can realize point-to-point audio-video communication, and provides a data transmission and signaling mechanism, so that real-time voice communication can be realized.
However, the existing implementation method has at least the following problems in practical application:
1. some clients do not support webtc technology;
2. real-time voice calls require a fast transfer rate and low delay. However, the conventional real-time voice transmission from the client to the server is often increased due to the influence of network delay, audio encoding and decoding delay and other factors, so that the real-time performance and fluency of the call are affected.
Disclosure of Invention
The invention aims to provide a WASM-based real-time voice communication method, a WASM-based real-time voice communication terminal and a camera, which are used for solving the problem of higher voice transmission delay in the prior art.
In order to achieve the above purpose, the present invention adopts the following technical scheme:
In a first aspect, the present invention provides a method for real-time voice communication based on WASM, applied to a client terminal, the method comprising:
Collecting original audio data;
encoding the original audio data based on a preset function instance to obtain an audio stream in a preset format;
Generating a receiving control request of an audio stream with a preset format;
The method comprises the steps of sending an audio stream with a preset format to a streaming media server and sending a receiving control request to a control platform, wherein the control platform is used for generating an audio stream pulling instruction according to the receiving control request, sending the audio stream pulling instruction to a camera based on SIP signaling, and the camera is used for pulling the audio stream with the preset format uploaded by a client terminal in the streaming media server based on the audio stream pulling instruction.
Preferably, the data transmission protocol for transmitting the audio stream in the preset format to the streaming media server is an RTMP protocol.
Preferably, collecting the raw audio data comprises:
When an audio acquisition request is initiated, loading WASM modules;
after the WASM module is loaded, the raw audio data is acquired based on the WASM module.
Preferably, the encoding of the original audio data based on the preset function instance to obtain an audio stream in a preset format includes:
invoking a preset function instance based on WASM modules;
encoding the original audio data by using a preset function instance to obtain an audio array;
compiling the audio array to obtain an audio stream in a preset format.
Preferably, the preset format is a G711A format.
Preferably, the control platform is further configured to: and simultaneously sending the audio stream pulling instruction to at least two cameras based on the SIP signaling.
In a second aspect, the present invention provides a WASM-based real-time voice communication method, which is applied to a camera, and the method includes:
Responding to an audio stream pulling instruction of a control platform, and pulling an audio stream with a preset format uploaded by a client terminal in a streaming media server, wherein the audio stream pulling instruction of the control platform is generated according to a receiving control request of the client terminal, and the audio stream with the preset format is an audio stream with the preset format obtained by encoding acquired original audio data by the client terminal;
Decoding an audio stream in a preset format to obtain decoded audio data;
and playing the decoded audio data.
Preferably, the data transmission protocol of the audio stream with the preset format pulled from the streaming media server is an RTMP protocol.
In a third aspect, the present invention provides a client terminal, configured to implement the foregoing WASM-based real-time voice communication method, where the terminal includes:
the data acquisition module is used for acquiring original audio data;
The encoding module is used for encoding the original audio data based on a preset function instance to obtain an audio stream in a preset format;
the request generation module is used for generating a receiving control request of the audio stream in a preset format;
The sending module is used for sending the audio stream with the preset format to the streaming media server and sending the receiving control request to the control platform, and the control platform is used for generating an audio stream pulling instruction according to the receiving control request and sending the audio stream pulling instruction to the camera based on the SIP signaling.
In a fourth aspect, the present invention provides a camera, configured to implement the foregoing WASM-based real-time voice communication method, where the camera includes:
The system comprises a pulling module, a receiving module and a receiving module, wherein the pulling module is used for responding to an audio stream pulling instruction of a control platform and pulling an audio stream with a preset format uploaded by a client terminal in a streaming media server, wherein the audio stream pulling instruction of the control platform is generated according to a receiving control request of the client terminal, and the audio stream with the preset format is an audio stream with the preset format obtained by encoding acquired original audio data by the client terminal;
The decoding module is used for decoding the audio stream with the preset format to obtain decoded audio data;
And the playing module is used for playing the decoded audio data.
The beneficial effects are that:
1. The communication method can complete the processing of the audio data more quickly, and has lower audio transmission delay;
2. The communication method can be operated on any browser, and has good portability and compatibility.
Drawings
The accompanying drawings are included to provide a further understanding of embodiments of the invention and are incorporated in and constitute a part of this specification, illustrate embodiments of the invention and together with the description serve to explain, without limitation, the embodiments of the invention. In the drawings:
FIG. 1 is a flow chart of a WASM-based real-time voice communication method provided by one embodiment of the present invention;
FIG. 2 is a flow chart of a WASM-based real-time voice communication method provided by an alternative embodiment of the present invention;
FIG. 3 is a block diagram of a client terminal provided by an alternative embodiment of the present invention;
FIG. 4 is a block diagram of a camera provided by an alternative embodiment of the present invention;
Fig. 5 is a diagram of a WASM-based real-time voice communication system provided in an alternative embodiment of the present invention.
Detailed Description
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the present invention will be briefly described below with reference to the accompanying drawings and the description of the embodiments or the prior art, and it is obvious that the following description of the structure of the drawings is only some embodiments of the present invention, and other drawings can be obtained according to these drawings without inventive effort to a person skilled in the art. It should be noted that the description of these examples is for aiding in understanding the present invention, but is not intended to limit the present invention.
Example 1
Fig. 1 is a flowchart of a method for real-time voice communication based on WASM according to one embodiment of the present invention, as shown in fig. 1, this embodiment provides a method for real-time voice communication based on WASM, which is applied to a client terminal, and the method includes:
Step S101: raw audio data is collected.
In this embodiment, a voice collection function is integrated on the client terminal, for example, a voice collection chip is integrated, and a browser of the client terminal collects voice information on site by acquiring voice collection rights.
Specifically, collecting raw audio data includes:
Step a01: when an audio acquisition request is initiated, loading WASM modules;
step a02: after the WASM module is loaded, the raw audio data is acquired based on the WASM module.
In this embodiment, when the client terminal needs to collect the on-site voice information, the processor of the client terminal generates an audio collection request, and sends the audio collection request of the processor to the voice collection chip, and the voice collection chip performs voice collection.
In this embodiment, the WASM module is a WebAssembly (virtual instruction set architecture) module, and when the client terminal initiates an audio acquisition request, the WASM module configured on the client terminal is loaded at the same time, and after the loading of the WASM module is completed, the WASM module is used to obtain original audio data (on-site voice information output by the voice acquisition chip).
Step S102: and encoding the original audio data based on the preset function instance to obtain an audio stream in a preset format.
Specifically, the method for encoding the original audio data based on the preset function instance to obtain an audio stream in a preset format includes:
Step b01: invoking a preset function instance based on WASM modules;
step b02: encoding the original audio data by using a preset function instance to obtain an audio array;
step b03: compiling the audio array to obtain an audio stream in a preset format.
In this embodiment, the preset format of the audio stream is the G711A format, that is, the WASM module is used to encode the original audio data, so as to obtain the audio stream in the G711A format.
Step S103: a reception control request of an audio stream of a preset format is generated.
Step S104: the method comprises the steps of sending an audio stream with a preset format to a streaming media server and sending a receiving control request to a control platform, wherein the control platform is used for generating an audio stream pulling instruction according to the receiving control request, sending the audio stream pulling instruction to a camera based on SIP signaling, and the camera is used for pulling the audio stream with the preset format uploaded by a client terminal in the streaming media server based on the audio stream pulling instruction.
In this embodiment, the browser itself does not have a function of establishing communication, and at this time, the execution of the communication task needs to be controlled by the control platform; the control platform of the embodiment can provide a series of security measures such as data encryption, identity verification and the like, and can better ensure the transmission security of the audio; meanwhile, the audio content can be effectively prevented from being intercepted or tampered illegally; the control platform can also perform centralized management and control on all browser streaming (pulling audio streams) activities, which means that an administrator can easily monitor, configure and optimize the streaming process to meet different requirements and scenes; and the control platform typically provides flexible configuration options that allow an administrator to adjust the pull parameters, such as code rate, resolution, frame rate, etc., according to actual needs. In addition, with the progress of technology and the change of user demands, the platform can easily expand new functions and services to meet the ever-increasing demands; secondly, the high quality and fluency of the audio can be ensured by controlling the browser stream pulling controlled by the platform, and the platform can perform intelligent processing on the audio stream, such as self-adaptive code rate adjustment, denoising, sharpening and the like, so as to improve the audio quality and user experience; and platforms typically support a variety of browsers and devices, ensuring that users can obtain a consistent pull experience on any platform. This helps to expand the user base and increase the popularity of applications. Therefore, the browser streaming controlled by the control platform has obvious advantages in the aspects of safety, centralized management, flexibility, video quality, compatibility and the like, and provides a better solution for developers and users.
In this embodiment, the data transmission protocol for transmitting the audio stream with the preset format to the streaming media server is an RTMP protocol, that is, the communication protocol between the client terminal and the streaming media server is an RTMP protocol, where the RTMP protocol (real-time message transmission protocol) is a protocol for transmitting audio, video and real-time data, and the RTMP protocol has higher instantaneity, flexibility and stability.
In this embodiment, before sending the audio stream in the preset format to the streaming media server, the method further includes: the client terminal is connected to the streaming media server by initiating an RTMP so as to establish an RTMP channel between the client terminal and the streaming media server; once the connection channel is established, the client terminal and the streaming server can send and receive data through the RTMP channel.
In this embodiment, the control platform is used to send the audio stream pulling instruction to the camera, and the camera starts to pull the audio stream in the streaming media server after receiving the audio stream pulling instruction, so as to send the collected audio data of the client terminal to the camera in real time.
As a further optimization of this embodiment, the control platform is further configured to: and simultaneously sending the audio stream pulling instruction to at least two cameras based on the SIP signaling so as to realize one-to-many real-time voice playing.
In this embodiment, the WebAssembly technology performs audio compilation, which has a faster execution speed and lower memory occupation than JavaScript, webAssembly, so that it becomes an advantageous choice in a specific scenario, especially in applications that process large amounts of data or require high performance.
In terms of audio transmission, the advantages of employing WebAssembly compilation mainly include:
1. Performance optimization: webAssembly performs faster than JavaScript, which means that computationally intensive tasks such as audio decoding, filtering or analysis during audio processing can be done faster;
2. Reducing delay: webAssembly allow for reduced audio processing and latency of transmission, which is critical for real-time audio transmission or applications requiring low latency;
3. And (3) reducing power consumption: webAssembly is low in memory occupation, and can reduce the burden of a processor, so that the power consumption is reduced, which is particularly important for mobile equipment and battery-powered equipment;
4. Cross-platform compatibility: webAssembly as a cross-platform binary format, can run on any browser supporting WebAssembly, which makes audio transmission solutions compiled based on WebAssembly better portability and compatibility;
5. safety: webAssembly is a compiled code format, which is more difficult to be maliciously utilized than JavaScript, thereby improving the security of the audio transmission process.
Example two
Fig. 2 is a flowchart of a WASM-based real-time voice communication method according to an alternative embodiment of the present invention, as shown in fig. 2, the present embodiment provides a WASM-based real-time voice communication method, which is applied to a camera, and the method includes:
step S201: responding to an audio stream pulling instruction of a control platform, and pulling an audio stream with a preset format uploaded by a client terminal in a streaming media server, wherein the audio stream pulling instruction of the control platform is generated according to a receiving control request of the client terminal, and the audio stream with the preset format is an audio stream with the preset format obtained by encoding acquired original audio data by the client terminal;
step S202: decoding an audio stream in a preset format to obtain decoded audio data;
step S203: and playing the decoded audio data.
In this embodiment, after receiving an audio stream pulling instruction of the control platform, the camera starts to pull an audio stream in a preset format in the streaming media server, that is, an audio stream in a G711A format, and then decodes the audio stream in the G711A format to obtain decoded audio data, where the decoded audio data is original audio data, and the camera plays the original audio.
In this embodiment, the camera adopts an IPC camera, and the camera has a built-in codec module, and decodes the audio stream in the G711A format through the codec module to obtain decoded audio data.
In this embodiment, the data transmission protocol of pulling the audio stream with the preset format from the streaming media server is an RTMP protocol, that is, the communication protocol between the camera and the streaming media server is an RTMP protocol, and in this embodiment, before pulling the audio stream with the preset format uploaded by the client terminal in the streaming media server, the method further includes: the camera is connected to the streaming media server by initiating an RTMP so as to establish an RTMP channel between the camera and the streaming media server; once the connection channel is established, the camera and the streaming server can send and receive data through the RTMP channel.
Example III
Fig. 3 is a block diagram of a client terminal according to an alternative embodiment of the present invention, and as shown in fig. 3, this embodiment provides a client terminal for implementing a method for implementing a real-time voice communication based on WASM in the first embodiment, where the terminal includes:
the data acquisition module is used for acquiring original audio data;
The encoding module is used for encoding the original audio data based on a preset function instance to obtain an audio stream in a preset format;
the request generation module is used for generating a receiving control request of the audio stream in a preset format;
The sending module is used for sending the audio stream with the preset format to the streaming media server and sending the receiving control request to the control platform, and the control platform is used for generating an audio stream pulling instruction according to the receiving control request and sending the audio stream pulling instruction to the camera based on the SIP signaling.
Example IV
Fig. 4 is a block diagram of a camera provided in an alternative embodiment of the present invention, as shown in fig. 4, and this embodiment provides a camera for implementing a WASM-based real-time voice communication method in the second embodiment, where the camera includes:
The system comprises a pulling module, a receiving module and a receiving module, wherein the pulling module is used for responding to an audio stream pulling instruction of a control platform and pulling an audio stream with a preset format uploaded by a client terminal in a streaming media server, wherein the audio stream pulling instruction of the control platform is generated according to a receiving control request of the client terminal, and the audio stream with the preset format is an audio stream with the preset format obtained by encoding acquired original audio data by the client terminal;
The decoding module is used for decoding the audio stream with the preset format to obtain decoded audio data;
And the playing module is used for playing the decoded audio data.
Example five
Fig. 5 is a diagram of a WASM-based real-time voice communication system according to an alternative embodiment of the present invention, as shown in fig. 5, the present embodiment provides a WASM-based real-time voice communication system, where the system includes: the client terminal, the control platform, the streaming media server and the camera of the fourth embodiment;
The client terminal is respectively in communication connection with the control platform and the streaming media server, the communication protocol between the client terminal and the streaming media server is RTMP protocol, each camera is respectively in communication connection with the control platform and the streaming media server, and the communication protocol between the cameras and the streaming media server is RTMP protocol.
The client terminal of the present embodiment is configured to: collecting original audio data; encoding the original audio data based on a preset function instance to obtain an audio stream in a preset format; generating a receiving control request of an audio stream with a preset format; and sending the audio stream with the preset format to the streaming media server and sending the receiving control request to the control platform.
The control platform of the present embodiment is configured to: and generating an audio stream pulling instruction according to the received control request, and simultaneously transmitting the audio stream pulling instruction to at least one camera based on the SIP signaling.
The camera of this embodiment is used for: responding to an audio stream pulling instruction of a control platform, and pulling an audio stream with a preset format uploaded by a client terminal in a streaming media server, wherein the audio stream pulling instruction of the control platform is generated according to a receiving control request of the client terminal, and the audio stream with the preset format is an audio stream with the preset format obtained by encoding acquired original audio data by the client terminal; decoding an audio stream in a preset format to obtain decoded audio data; and playing the decoded audio data.
The communication system of the embodiment can complete the processing of the audio data more quickly, and has lower audio transmission delay; and can run on any browser, and has good portability and compatibility.
It will be appreciated by those skilled in the art that embodiments of the present application may be provided as a method, system, or computer program product. Accordingly, the present application may take the form of an entirely hardware embodiment, an entirely software embodiment or an embodiment combining software and hardware aspects. Furthermore, the present application may take the form of a computer program product embodied on one or more computer-usable storage media (including, but not limited to, disk storage, CD-ROM, optical storage, and the like) having computer-usable program code embodied therein.
The present application is described with reference to flowchart illustrations and/or block diagrams of methods, apparatus (systems) and computer program products according to embodiments of the application. It will be understood that each flow and/or block of the flowchart illustrations and/or block diagrams, and combinations of flows and/or blocks in the flowchart illustrations and/or block diagrams, can be implemented by computer program instructions. These computer program instructions may be provided to a processor of a general purpose computer, special purpose computer, embedded processor, or other programmable data processing apparatus to produce a machine, such that the instructions, which execute via the processor of the computer or other programmable data processing apparatus, create means for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.
The foregoing is merely exemplary of the present application and is not intended to limit the present application. Various modifications and variations of the present application will be apparent to those skilled in the art. Any modification, equivalent replacement, improvement, etc. which come within the spirit and principles of the application are to be included in the scope of the claims of the present application.
Claims (10)
1. A WASM-based real-time voice communication method applied to a client terminal, the method comprising:
Collecting original audio data;
encoding the original audio data based on a preset function instance to obtain an audio stream in a preset format;
Generating a receiving control request of an audio stream with a preset format;
The method comprises the steps of sending an audio stream with a preset format to a streaming media server and sending a receiving control request to a control platform, wherein the control platform is used for generating an audio stream pulling instruction according to the receiving control request, sending the audio stream pulling instruction to a camera based on SIP signaling, and the camera is used for pulling the audio stream with the preset format uploaded by a client terminal in the streaming media server based on the audio stream pulling instruction.
2. The WASM-based real-time voice communication method as claimed in claim 1, wherein the data transmission protocol for transmitting the audio stream of the preset format to the streaming server is an RTMP protocol.
3. The WASM-based real-time voice communication method as claimed in claim 1, wherein collecting raw audio data comprises:
When an audio acquisition request is initiated, loading WASM modules;
after the WASM module is loaded, the raw audio data is acquired based on the WASM module.
4. The WASM-based real-time voice communication method according to claim 3, wherein the encoding of the original audio data based on the preset function instance to obtain the audio stream in the preset format includes:
invoking a preset function instance based on WASM modules;
encoding the original audio data by using a preset function instance to obtain an audio array;
compiling the audio array to obtain an audio stream in a preset format.
5. The WASM-based real-time voice communication method according to claim 4, wherein the preset format is G711A format.
6. The WASM-based real-time voice communication method as claimed in claim 1, wherein the control platform is further configured to: and simultaneously sending the audio stream pulling instruction to at least two cameras based on the SIP signaling.
7. A WASM-based real-time voice communication method applied to a camera, the method comprising:
Responding to an audio stream pulling instruction of a control platform, and pulling an audio stream with a preset format uploaded by a client terminal in a streaming media server, wherein the audio stream pulling instruction of the control platform is generated according to a receiving control request of the client terminal, and the audio stream with the preset format is an audio stream with the preset format obtained by encoding acquired original audio data by the client terminal;
Decoding an audio stream in a preset format to obtain decoded audio data;
and playing the decoded audio data.
8. The WASM-based real-time voice communication method as claimed in claim 7, wherein the data transmission protocol of pulling the audio stream of the preset format from the streaming server is an RTMP protocol.
9. A client terminal for implementing the WASM-based real-time voice communication method of any one of claims 1-6, the terminal comprising:
the data acquisition module is used for acquiring original audio data;
The encoding module is used for encoding the original audio data based on a preset function instance to obtain an audio stream in a preset format;
the request generation module is used for generating a receiving control request of the audio stream in a preset format;
The sending module is used for sending the audio stream with the preset format to the streaming media server and sending the receiving control request to the control platform, and the control platform is used for generating an audio stream pulling instruction according to the receiving control request and sending the audio stream pulling instruction to the camera based on the SIP signaling.
10. A camera for implementing the WASM-based real-time voice communication method of claim 7 or 8, the camera comprising:
The system comprises a pulling module, a receiving module and a receiving module, wherein the pulling module is used for responding to an audio stream pulling instruction of a control platform and pulling an audio stream with a preset format uploaded by a client terminal in a streaming media server, wherein the audio stream pulling instruction of the control platform is generated according to a receiving control request of the client terminal, and the audio stream with the preset format is an audio stream with the preset format obtained by encoding acquired original audio data by the client terminal;
The decoding module is used for decoding the audio stream with the preset format to obtain decoded audio data;
And the playing module is used for playing the decoded audio data.
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