preface

AnyRTC SDK was updated in December with many new features. To provide more convenient development environment for developers.

For more detailed documents and historical updates of anyRTC SDK, you can visit anyRTC official website – Developer Center – Documentation Center.

This month’s updates include:

  • RTC added self-rendering
  • encryption
  • The watermark
  • Custom video stream parameter Settings
  • Uni-app cross-platform SDK real-time audio and video call plug-in
  • Uni-app Cross-platform SDK real-time messaging plug-in
  • Audio music adaptation mode

Here is a brief introduction of each function for you;

AnyRTC custom video self rendering

Self-rendering: Self-rendering, also known as custom rendering, refers to the process of self-rendering by developers after obtaining raw audio and video data from the SDK.

An external renderer can be used to render audio and video data when the default audio and video module does not meet development requirements. Such as:

  • The obtained raw audio and video data is passed to other audio and video rendering engines.
  • Need to customize audio and video rendering methods, such as custom render animation, etc.
  • When the default renderer is occupied by other services, external renderers are required to render the original audio and video data to avoid conflicts between the audio and video services.

Function introduction:

In the process of real-time video transmission, anyRTC SDK usually starts the default video module for collection and rendering. In the following scenarios, you may find that the default video module does not meet your development requirements:

  • App already has its own video module
  • Want to use non-camera captured video sources, such as screen recording data
  • Need to use a custom beauty library or have a pre – processing library
  • Some video capture devices are exclusively used by the system. Flexible device management policies are required to avoid conflicts with other services

Based on this, the anyRTC SDK supports the use of custom video sources or renderers to achieve related scenes.

Data flow diagram

Media stream encryption

Functional description

During real-time audio and video interaction, developers need to encrypt media streams to ensure user data security.

Streaming media encryption mechanism

You can see a pattern that these streaming protocols all have to break the video stream into sequential chunks that are then transmitted, but the size of the chunks is different from the format of the video container.

The core of streaming media encryption technology is to use symmetric encryption algorithm for each piece of video, and decrypt the video on the server side and the client side, so that the user who passes the permission verification can get the key to decrypt a piece of video.

At present, anyRTC media stream encryption has corresponding API interface, which can be customized to enable and disable, convenient for developers to use.

The watermark

Functional description

Some users worry about their live video being used illegally or need to show their brand, they need to add watermarks on the video, such as company logo, time stamp or specific text information, for the purpose of anti-counterfeiting, copyright declaration, publicity or recording.

AnyRTC provides two ways to add watermarks on the local end and the server end. There are different interfaces for developers to choose from.

Currently anyRTC can add a static image watermark to a video, using a local PNG image as the watermark.

Set the watermark size and location

To add any watermark, you must set the horizontal position, vertical position, width, and height of the watermark using offsetX, offsetY, wmWidth, and wmHeight.

Custom video stream parameter Settings

Functional description

Before or after joining the channel, each streaming end can call open dual-stream mode. After this function is enabled, the SDK sends an additional video stream with low resolution and low bit rate while sending video streams. Among them, the original video stream is also called stream, and the stream with lower resolution and bit rate is called small stream.

The SDK automatically sets the default video properties of the stream based on the stream’s video properties.

AnyRTC custom stream parameters, previously the default, can now be customized after more flexible and convenient.

Uni-app cross-platform SDK real-time audio and video call plug-in

Functional description

Can create a room, anchor end and visitors end, can be multi-person video call, also can point to point call.

When entering the host terminal, you can make video calls for single or multiple users by entering the same room number.

The tourist end is as shown in the figure:

The visitor side can only be viewed and will not be displayed on the home screen.



The anchor end is as shown in the figure:



Support platform

  • Android
  • ios

Application scenarios

1, point-to-point voice chat, support 720P, 1080P hd quality.

2, voice group chat Online group chat, a maximum of 300 people in a single room can be online at the same time, and a maximum of 50 people can open the camera at the same time.

3. In the low-delay mode of live video broadcast, 100,000 viewers are supported to play simultaneously, and the playback delay is as low as 300ms. In CDN bypass mode, the number of viewers is unlimited.

4. Video Link supports the interaction between anchor and audience video link, and live broadcast across rooms (across live broadcast rooms) PK.

GitHub download: github.com/anyRTC/uni-…

Uni-app Cross-platform SDK real-time messaging plug-in

Functional description

Uni-app cross-platform real-time messaging SDK plug-in, including point-to-point messaging, channel messaging, call invitation and other functions. Under the same channel, users can receive and send messages. From the status bar below, you can view the status changes in the channel, including personnel updates and message updates.

As is shown in



Support platform

  • Android
  • ios

Application scenarios

1. Online Education In online education scenarios, applicable to large and small classes, students can send messages to communicate with teachers in real time.

2. Entertainment Live Broadcast In the live broadcast environment, the audience can send bullet screen in real time to interact with the anchor.

3. Intercom In railway or transportation scenarios, employees in the same company and department can communicate with each other on the same channel, greatly improving work efficiency and reducing communication costs.

4, voice group chat in the voice chat environment can also support message communication.

GitHub download: github.com/anyRTC/uni-…

Audio music adaptation mode

In order to ensure that the voice can be transmitted to the user more clearly and completely during voice chat or live broadcast, anyRTC has upgraded the audio quality from the two aspects of “supporting dual-channel audio playback” and “improving the sound quality in music mode”.

  • Supports dual-channel audio playback

At present, anyRTC SDK can guarantee dual-channel playback, so that the host end can transmit high-quality audio to users during live broadcast, so that users can experience more professional music enjoyment.

  • Improved sound quality in music mode

AnyRTC has always been committed to bringing a better audio and video experience to users, so this update specifically improves the sound quality in music mode.

Can be applied in high-pitched language chat room, live show, sound quality priority and better professional peripherals support, game open black.

The above is the main content of SDK iteration this month, more detailed documents and anyRTC SDK historical updates, you can go to “anyRTC official website – Developer Center – Documentation Center” to view.