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Carte de Schillerkiez
13 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
Autres articles (78)
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Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
MediaSPIP Player : problèmes potentiels
22 février 2011, parLe lecteur ne fonctionne pas sur Internet Explorer
Sur Internet Explorer (8 et 7 au moins), le plugin utilise le lecteur Flash flowplayer pour lire vidéos et son. Si le lecteur ne semble pas fonctionner, cela peut venir de la configuration du mod_deflate d’Apache.
Si dans la configuration de ce module Apache vous avez une ligne qui ressemble à la suivante, essayez de la supprimer ou de la commenter pour voir si le lecteur fonctionne correctement : /** * GeSHi (C) 2004 - 2007 Nigel McNie, (...) -
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
Sur d’autres sites (4926)
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Documentation #3027 : Squelette Spip.net
18 novembre 2013, par Arnaud Dupin de BeyssatBonjour
Merci de la réponse.
Il s’agit notammen t du menu latéral de la page
http://www.spip.net/fr_rubrique135.html et des suivantes (filrtres,
critères ; etc.)
sans doute associéà<a class="external" href="http://www.spip.net/">www.spip.net</a><br /> /squelettes<br /> /branches<br /> /2008<br /> /inc-rubriques.html
Merci
ADBLe 16/07/2013 10:35, redmine@spip.org a écrit :
La demande #3027 a été mise à jour par b b.
Salut, de quelle page du site s’agit-il ?
Evolution #3027 : Squelette Spip.net
- Auteur : Arnaud Dupin de Beyssat
- Statut : Nouveau
- Priorité : Normal
- Assigné à :
- Catégorie :
- Version cible :
- Resolution :
Bonjour
Serait-il possible de passer la liste des rubriques du menu latéral
ordonné en alphabétique ? Cela faciliterait les recherches de balises, etc.
Actuellement, les lignes sont :
12
13 id_rubriqueid_rubrique=#ENVid_rubrique par num titre !par date
doublons>
14 #TITRE
15A la ligne 13, passer en par nom ai lieu de par num titre !par date
Merci
Vous recevez ce mail car vous êtes impliqués sur ce projet.
Pour changer les préférences d’envoi de mail, allez sur
http://core.spip.org/my/account -
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 ?








-
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);
 }
 };