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Sur d’autres sites (4921)
-
How do i play an HLS stream when playlist.m3u8 file is constantly being updated ?
3 janvier 2021, par Adnan AhmedI am using MediaRecorder to record chunks of my live video in webm format from MediaStream and converting these chunks to .ts files on the server using ffmpeg and then updating my playlist.m3u8 file with this code :


function generateM3u8Playlist(fileDataArr, playlistFp, isLive, cb) {
 var durations = fileDataArr.map(function(fd) {
 return fd.duration;
 });
 var maxT = maxOfArr(durations);

 var meta = [
 '#EXTM3U',
 '#EXT-X-VERSION:3',
 '#EXT-X-MEDIA-SEQUENCE:0',
 '#EXT-X-ALLOW-CACHE:YES',
 '#EXT-X-TARGETDURATION:' + Math.ceil(maxT),
 ];

 fileDataArr.forEach(function(fd) {
 meta.push('#EXTINF:' + fd.duration.toFixed(2) + ',');
 meta.push(fd.fileName2);
 });

 if (!isLive) {
 meta.push('#EXT-X-ENDLIST');
 }

 meta.push('');
 meta = meta.join('\n');

 fs.writeFile(playlistFp, meta, cb);
}



Here
fileDataArr
holds information for all the chunks that have been created.

After that i use this code to create a hls server :


var runStreamServer = (function(streamFolder) {
 var executed = false;
 return function(streamFolder) {
 if (!executed) {
 executed = true;
 var HLSServer = require('hls-server')
 var http = require('http')

 var server = http.createServer()
 var hls = new HLSServer(server, {
 path: '/stream', // Base URI to output HLS streams
 dir: 'C:\\Users\\Work\\Desktop\\live-stream\\webcam2hls\\videos\\' + streamFolder // Directory that input files are stored
 })
 console.log("We are going to stream from folder:" + streamFolder);
 server.listen(8000);
 console.log('Server Listening on Port 8000');
 }
 };
})();



The problem is that if i stop creating new chunks and then use the hls server link :

http://localhost:8000/stream/playlist.m3u8
then the video plays in VLC but if i try to play during the recording it keeps loading the file but does not play. I want it to play while its creating new chunks and updating playlist.m3u8. The quirk ingenerateM3u8Playlist
function is that it adds'#EXT-X-ENDLIST'
to the playlist file after i have stopped recording.
The software is still in production so its a bit messy code. Thank you for any answers.

The client side that generates blobs is as follows :


var mediaConstraints = {
 video: true,
 audio:true
 };
navigator.getUserMedia(mediaConstraints, onMediaSuccess, onMediaError);
function onMediaSuccess(stream) {
 console.log('will start capturing and sending ' + (DT / 1000) + 's videos when you press start');
 var mediaRecorder = new MediaStreamRecorder(stream);

 mediaRecorder.mimeType = 'video/webm';

 mediaRecorder.ondataavailable = function(blob) {
 var count2 = zeroPad(count, 5);
 // here count2 just creates a blob number 
 console.log('sending chunk ' + name + ' #' + count2 + '...');
 send('/chunk/' + name + '/' + count2 + (stopped ? '/finish' : ''), blob);
 ++count;
 };
 }
// Here we have the send function which sends our blob to server:
 function send(url, blob) {
 var xhr = new XMLHttpRequest();
 xhr.open('POST', url, true);

 xhr.responseType = 'text/plain';
 xhr.setRequestHeader('Content-Type', 'video/webm');
 //xhr.setRequestHeader("Content-Length", blob.length);

 xhr.onload = function(e) {
 if (this.status === 200) {
 console.log(this.response);
 }
 };
 xhr.send(blob);
 }



The code that receives the XHR request is as follows :


var parts = u.split('/');
 var prefix = parts[2];
 var num = parts[3];
 var isFirst = false;
 var isLast = !!parts[4];

 if ((/^0+$/).test(num)) {
 var path = require('path');
 shell.mkdir(path.join(__dirname, 'videos', prefix));
 isFirst = true;
 }

 var fp = 'videos/' + prefix + '/' + num + '.webm';
 var msg = 'got ' + fp;
 console.log(msg);
 console.log('isFirst:%s, isLast:%s', isFirst, isLast);

 var stream = fs.createWriteStream(fp, { encoding: 'binary' });
 /*stream.on('end', function() {
 respond(res, ['text/plain', msg]);
 });*/

 //req.setEncoding('binary');

 req.pipe(stream);
 req.on('end', function() {
 respond(res, ['text/plain', msg]);

 if (!LIVE) { return; }

 var duration = 20;
 var fd = {
 fileName: num + '.webm',
 filePath: fp,
 duration: duration
 };
 var fileDataArr;
 if (isFirst) {
 fileDataArr = [];
 fileDataArrs[prefix] = fileDataArr;
 } else {
 var fileDataArr = fileDataArrs[prefix];
 }
 try {
 fileDataArr.push(fd);
 } catch (err) {
 fileDataArr = [];
 console.log(err.message);
 }
 videoUtils.computeStartTimes(fileDataArr);

 videoUtils.webm2Mpegts(fd, function(err, mpegtsFp) {
 if (err) { return console.error(err); }
 console.log('created %s', mpegtsFp);

 var playlistFp = 'videos/' + prefix + '/playlist.m3u8';

 var fileDataArr2 = (isLast ? fileDataArr : lastN(fileDataArr, PREV_ITEMS_IN_LIVE));

 var action = (isFirst ? 'created' : (isLast ? 'finished' : 'updated'));

 videoUtils.generateM3u8Playlist(fileDataArr2, playlistFp, !isLast, function(err) {
 console.log('playlist %s %s', playlistFp, (err ? err.toString() : action));
 });
 });


 runStreamServer(prefix);
 }



-
webrtc to rtmp send video from camera to rtmp link
14 avril 2024, par Leo-Mahendrai cant send the video from webrtc which is converted to bufferd data for every 10seconds and send to server.js where it takes it via websockets and convert it to flv format using ffmpeg.


i am trying to send it to rtmp server named restreamer for start, here i tried to convert the buffer data and send it to rtmp link using ffmpeg commands, where i initially started to suceesfully save the file from webrtc to mp4 format for a duration of 2-3 minute.


after i tried to use webrtc to send video data for every 10 seconds and in server i tried to send it to rtmp but i cant send it, but i can see the connection of rtmp url and server is been taken place but i cant see the video i can see the logs in rtmp server as


2024-04-14 12:35:45 ts=2024-04-14T07:05:45Z level=INFO component="RTMP" msg="no streams available" action="INVALID" address=":1935" client="172.17.0.1:37700" path="/3d30c5a9-2059-4843-8957-da963c7bc19b.stream" who="PUBLISH"
2024-04-14 12:35:45 ts=2024-04-14T07:05:45Z level=INFO component="RTMP" msg="no streams available" action="INVALID" address=":1935" client="172.17.0.1:37716" path="/3d30c5a9-2059-4843-8957-da963c7bc19b.stream" who="PUBLISH"
2024-04-14 12:35:45 ts=2024-04-14T07:05:45Z level=INFO component="RTMP" msg="no streams available" action="INVALID" address=":1935" client="172.17.0.1:37728" path="/3d30c5a9-2059-4843-8957-da963c7bc19b.stream" who="PUBLISH" 



my frontend code


const handleSendVideo = async () => {
 console.log("start");
 
 if (!ws) {
 console.error('WebSocket connection not established.');
 return;
 }
 
 try {
 const videoStream = await navigator.mediaDevices.getUserMedia({ video: true });
 const mediaRecorder = new MediaRecorder(videoStream);
 
 const requiredFrameSize = 460800;
 const frameDuration = 10 * 1000; // 10 seconds in milliseconds
 
 mediaRecorder.ondataavailable = async (event) => {
 if (ws.readyState !== WebSocket.OPEN) {
 console.error('WebSocket connection is not open.');
 return;
 }
 
 if (event.data.size > 0) {
 const arrayBuffer = await event.data.arrayBuffer();
 const uint8Array = new Uint8Array(arrayBuffer);
 
 const width = videoStream.getVideoTracks()[0].getSettings().width;
 const height = videoStream.getVideoTracks()[0].getSettings().height;
 
 const numFrames = Math.ceil(uint8Array.length / requiredFrameSize);
 
 for (let i = 0; i < numFrames; i++) {
 const start = i * requiredFrameSize;
 const end = Math.min((i + 1) * requiredFrameSize, uint8Array.length);
 let frameData = uint8Array.subarray(start, end);
 
 // Pad or trim the frameData to match the required size
 if (frameData.length < requiredFrameSize) {
 // Pad with zeros to reach the required size
 const paddedData = new Uint8Array(requiredFrameSize);
 paddedData.set(frameData, 0);
 frameData = paddedData;
 } else if (frameData.length > requiredFrameSize) {
 // Trim to match the required size
 frameData = frameData.subarray(0, requiredFrameSize);
 }
 
 const dataToSend = {
 buffer: Array.from(frameData), // Convert Uint8Array to array of numbers
 width: width,
 height: height,
 pixelFormat: 'yuv420p',
 mode: 'SendRtmp'
 };
 
 console.log("Sending frame:", i);
 ws.send(JSON.stringify(dataToSend));
 }
 }
 };
 
 // Start recording and send data every 10 seconds
 mediaRecorder.start(frameDuration);
 
 console.log("MediaRecorder started.");
 } catch (error) {
 console.error('Error accessing media devices or starting recorder:', error);
 }
 };



and my backend


wss.on('connection', (ws) => {
 console.log('WebSocket connection established.');

 ws.on('message', async (data) => {
 try {
 const parsedData = JSON.parse(data);

 if (parsedData.mode === 'SendRtmp' && Array.isArray(parsedData.buffer)) {
 const { buffer, pixelFormat, width, height } = parsedData;
 const bufferArray = Buffer.from(buffer);

 await sendRtmpVideo(bufferArray, pixelFormat, width, height);
 } else {
 console.log('Received unknown or invalid mode or buffer data');
 }
 } catch (error) {
 console.error('Error parsing WebSocket message:', error);
 }
 });

 ws.on('close', () => {
 console.log('WebSocket connection closed.');
 });
 });
 const sendRtmpVideo = async (frameBuffer, pixelFormat, width, height) => {
 console.log("ffmpeg data",frameBuffer)
 try {
 const ratio = `${width}x${height}`;
 const ffmpegCommand = [
 '-re',
 '-f', 'rawvideo',
 '-pix_fmt', pixelFormat,
 '-s', ratio,
 '-i', 'pipe:0',
 '-c:v', 'libx264',
 '-preset', 'fast', // Specify the preset for libx264
 '-b:v', '3000k', // Specify the video bitrate
 '-loglevel', 'debug',
 '-f', 'flv',
 // '-flvflags', 'no_duration_filesize', 
 RTMPLINK
 ];


 const ffmpeg = spawn('ffmpeg', ffmpegCommand);

 ffmpeg.on('exit', (code, signal) => {
 if (code === 0) {
 console.log('FFmpeg process exited successfully.');
 } else {
 console.error(`FFmpeg process exited with code ${code} and signal ${signal}`);
 }
 });

 ffmpeg.on('error', (error) => {
 console.error('FFmpeg spawn error:', error);
 });

 ffmpeg.stderr.on('data', (data) => {
 console.error(`FFmpeg stderr: ${data}`);
 });

 ffmpeg.stdin.write(frameBuffer, (err) => {
 if (err) {
 console.error('Error writing to FFmpeg stdin:', err);
 } else {
 console.log('Data written to FFmpeg stdin successfully.');
 }
 ffmpeg.stdin.end(); // Close stdin after writing the buffer
 });
 } catch (error) {
 console.error('Error in sendRtmpVideo:', error);
 }
 };




-
Transcode webcam blob to RTMP using ffmpeg.wasm
29 novembre 2023, par hassan moradnezhadI'm trying transcode webcam blob data to a rtmp server from browser by using ffmpeg.wasm .

first, i create a MediaStream.

const stream = await navigator.mediaDevices.getUserMedia({
 video: true,
 });



then, i create a MediaRecorder.


const recorder = new MediaRecorder(stream, {mimeType: "video/webm; codecs:vp9"});
 recorder.ondataavailable = handleDataAvailable;
 recorder.start(0)



when data is available, i call a function called
handleDataAvailable
.

here is the function.

const handleDataAvailable = (event: BlobEvent) => {
 console.log("data-available");
 if (event.data.size > 0) {
 recordedChunksRef.current.push(event.data);
 transcode(event.data)
 }
 };



in above code, i use another function which called
transcode
it's goal is going to send data to rtmp server using useffmpeg.wasm
.

here it is.

const transcode = async (inputVideo: Blob | undefined) => {
 const ffmpeg = ffmpegRef.current;
 const fetchFileOutput = await fetchFile(inputVideo)
 ffmpeg?.writeFile('input.webm', fetchFileOutput)

 const data = await ffmpeg?.readFile('input.webm');
 if (videoRef.current) {
 videoRef.current.src =
 URL.createObjectURL(new Blob([(data as any)?.buffer], {type: 'video/webm'}));
 }

 // execute by node-media-server config 1
 await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c', 'copy', '-f', 'flv', "rtmp://localhost:1935/live/ttt"])

 // execute by node-media-server config 2
 // await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c:v', 'libx264', '-preset', 'veryfast', '-tune', 'zerolatency', '-c:a', 'aac', '-ar', '44100', '-f', 'flv', 'rtmp://localhost:1935/live/ttt']);

 // execute by stack-over-flow config 1
 // await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c:v', 'h264', '-c:a', 'aac', '-f', 'flv', "rtmp://localhost:1935/live/ttt"]);

 // execute by stack-over-flow config 2
 // await ffmpeg?.exec(['-i', 'input.webm', '-c:v', 'libx264', '-flags:v', '+global_header', '-c:a', 'aac', '-ac', '2', '-f', 'flv', "rtmp://localhost:1935/live/ttt"]);

 // execute by stack-over-flow config 3
 // await ffmpeg?.exec(['-i', 'input.webm', '-acodec', 'aac', '-ac', '2', '-strict', 'experimental', '-ab', '160k', '-vcodec', 'libx264', '-preset', 'slow', '-profile:v', 'baseline', '-level', '30', '-maxrate', '10000000', '-bufsize', '10000000', '-b', '1000k', '-f', 'flv', 'rtmp://localhost:1935/live/ttt']);

 }



after running app and start streaming, console logs are as below.


ffmpeg >>> ffmpeg version 5.1.3 Copyright (c) 2000-2022 the FFmpeg developers index.tsx:81:20
ffmpeg >>> built with emcc (Emscripten gcc/clang-like replacement + linker emulating GNU ld) 3.1.40 (5c27e79dd0a9c4e27ef2326841698cdd4f6b5784) index.tsx:81:20
ffmpeg >>> configuration: --target-os=none --arch=x86_32 --enable-cross-compile --disable-asm --disable-stripping --disable-programs --disable-doc --disable-debug --disable-runtime-cpudetect --disable-autodetect --nm=emnm --ar=emar --ranlib=emranlib --cc=emcc --cxx=em++ --objcc=emcc --dep-cc=emcc --extra-cflags='-I/opt/include -O3 -msimd128' --extra-cxxflags='-I/opt/include -O3 -msimd128' --disable-pthreads --disable-w32threads --disable-os2threads --enable-gpl --enable-libx264 --enable-libx265 --enable-libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libopus --enable-zlib --enable-libwebp --enable-libfreetype --enable-libfribidi --enable-libass --enable-libzimg index.tsx:81:20
ffmpeg >>> libavutil 57. 28.100 / 57. 28.100 index.tsx:81:20
ffmpeg >>> libavcodec 59. 37.100 / 59. 37.100 index.tsx:81:20
ffmpeg >>> libavformat 59. 27.100 / 59. 27.100 index.tsx:81:20
ffmpeg >>> libavdevice 59. 7.100 / 59. 7.100 index.tsx:81:20
ffmpeg >>> libavfilter 8. 44.100 / 8. 44.100 index.tsx:81:20
ffmpeg >>> libswscale 6. 7.100 / 6. 7.100 index.tsx:81:20
ffmpeg >>> libswresample 4. 7.100 / 4. 7.100 index.tsx:81:20
ffmpeg >>> libpostproc 56. 6.100 / 56. 6.100 index.tsx:81:20
ffmpeg >>> Input #0, matroska,webm, from 'input.webm': index.tsx:81:20
ffmpeg >>> Metadata: index.tsx:81:20
ffmpeg >>> encoder : QTmuxingAppLibWebM-0.0.1 index.tsx:81:20
ffmpeg >>> Duration: N/A, start: 0.000000, bitrate: N/A index.tsx:81:20
ffmpeg >>> Stream #0:0(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 4:3, 15.50 tbr, 1k tbn (default)



the problem is when
ffmpeg.wasm
try to execute the last command.

await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c', 'copy', '-f', 'flv', "rtmp://localhost:1935/live/ttt"])
.

it just calls aGET Request
, I will send further details about this request.

as u can see, i try to use lots of arg sample withffmpeg?.exec
, but non of them works.

the network tab in browser, after
ffmpeg.wasm
execute the command is as below.



it send a
GET request
tows://localhost:1935/

and nothing happened after that.

for backend, i use node-media-server and here is my output logs when
ffmpeg.wasm
trying to execute the args

11/28/2023 19:33:18 55301 [INFO] [rtmp disconnect] id=JL569YOF
[NodeEvent on doneConnect] id=JL569YOF args=undefined



at last here are my ques




- 

- how can i achive this option ?
- is it possible to share webcam to rtmp server ?