Programming VIP

MediaDevices.getUserMedia() will prompt the user to grant permission to use media input. Media input will generate a MediaStream containing the track of the requested media type. This stream can contain a video track (from hardware or virtual video sources, such as cameras, video capture devices, screen sharing services, etc.), an audio track ( ...

Introduction to webrtc Chapter 4 data channel 1, Introduction In the process of realizing end-to-end connection in webrtc, in addition to transmitting media stream, it can also transmit text, files, pictures and other data. It is also practical in IM scenarios, and does not need third-party message services, such as websocket, transparent mes ...

explain Because the code is not completely completed, the complete code is not put up. If necessary, you can email me and I will send the code to you. But the most important key parts have been posted. The picture shows receiving RTSP server video, receiving, playing, analyzing and forwarding flv. After the completion of the framework, one is ...

in front WebRTC server construction We have built the server environment required by WebRTC, mainly including three servers: Room server, signaling server and TURN penetration server. Later, we will learn how to use WebRTC to realize audio and video calls step by step. Today we will learn how to preview camera data using WebRTC. Here's the bo ...

stay WebRTC In the video processing pipeline of, the receiver buffer JitterBuffer is a key component: it is responsible for the out of order rearrangement and framing of RTP packets, RTP packet loss retransmission, requesting retransmission of key frames, estimating buffer delay and other functions. The buffer delay JitterDelay has an important ...

prefaceWith the influence of COVID-19, the demand for audio and video has explode in the past two years. In the field of audio and video, WebRTC can be said to be an inseparable treasure house, including the whole process of audio and video acquisition, encoding and decoding, transmission and rendering. This article mainly records the whole pro ...

brief introduction In the rr (receiver report) packet of rtcp, there are two fields, fraction lost and cumulative number of packets lost, which are used to indicate the packet loss rate and the total number of packets lost. This article will introduce the definition of these two values and the implementation in the chrome kernel. cause T ...

When using WebRTC, there are generally two ways to beautify the video: replacing the acquisition module in WebRTC and beautifying the video data. 1, Replace the acquisition module in WebRTC It is relatively simple to replace the acquisition module in WebRTC. Use GPUImageVideoCamera to replace the video acquisition in WebRTC, get the image aft ...

prefaceWith the influence of COVID-19, the demand for audio and video has explode in the past two years. In the field of audio and video, WebRTC can be said to be an inseparable treasure house, including the whole process of audio and video acquisition, encoding and decoding, transmission and rendering. This paper mainly records the whole proce ...

To realize audio visualization and achieve some cool effects, we need to use some methods AudioContext provided by Web Audio API. The AudioContext interface represents the audio processing diagram constructed by linked audio modules, and each module is represented by an AudioNode. The audio context controls the creation of the nodes it contai ...