http-live-streaming

来源： https://stackoverflow.com/questions/50628791/decrypt-m3u8-playlist-encrypted-with-aes-128-without-iv

来源： https://stackoverflow.com/questions/50628791/decrypt-m3u8-playlist-encrypted-with-aes-128-without-iv

来源： https://stackoverflow.com/questions/50628791/decrypt-m3u8-playlist-encrypted-with-aes-128-without-iv

来源： https://stackoverflow.com/questions/50628791/decrypt-m3u8-playlist-encrypted-with-aes-128-without-iv

来源： https://stackoverflow.com/questions/50628791/decrypt-m3u8-playlist-encrypted-with-aes-128-without-iv

来源： https://stackoverflow.com/questions/63772454/how-to-play-a-live-video-in-the-browser

问题 Currently I am working on an idea for a "Skype" like application. E.g people video call other people. Now I thought of running this on Microsoft Azure Media Services. However I'm not quite sure if this is possible. I found this video explaining how to make a simple stream from your computers webcam. What they do there is create a Channel (manually in the Azure environment) and stream to the ingest url . And the result can be seen in the publish url . So far so good. I understand this. But

问题 I'm trying to build a custom progress bar for a video player app in tvOS, and would like to show thumbnails of the video while the user scans the video. I'm using AVPlayer and Fairplay HLS to play remote video files. I've tried to do this using 2 methods. One with AVAssetImageGenerator 's copyCGImage , and the other with AVPlayerItemVideoOutput 's copyPixelBuffer method. Both return nil. When I tried with a local video file, the first method worked. Method 1: let imageGenerator =

问题 As a proof-of-concept I need to create a HTTP server which on GET request should start continuous stream of non-encoded/non-compressed audio data - WAV, PCM16. Let's assume the audio data are chunks of 4096 randomly generated mono audio samples @44.1kHz sampling rate. What should I put in the HTTP response header in order to browser on the other end start a player in its UI for the user to listen in realtime? I was reading about "Transfer-Encoding: chunked", "multipart", mimetype="audio/xwav"

问题 I'm trying to play video from websocket <video id="output" width="320" height="240" autoplay></video> <script> function sockets(buffer) { const socket = new WebSocket('wss://localhost:5002/ws') socket.onmessage = async function (event) { // event.data is a blob buffer.appendBuffer(new Uint8Array(event.data)) } } let ms = new MediaSource() let output = document.getElementById('output') output.src = URL.createObjectURL(ms) ms.onsourceopen = () => { let buffer = ms.addSourceBuffer('video/webm;