Recherche avancée
Autres articles (39)
-
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
-
Possibilité de déploiement en ferme
12 avril 2011, parMediaSPIP peut être installé comme une ferme, avec un seul "noyau" hébergé sur un serveur dédié et utilisé par une multitude de sites différents.
Cela permet, par exemple : de pouvoir partager les frais de mise en œuvre entre plusieurs projets / individus ; de pouvoir déployer rapidement une multitude de sites uniques ; d’éviter d’avoir à mettre l’ensemble des créations dans un fourre-tout numérique comme c’est le cas pour les grandes plate-formes tout public disséminées sur le (...)
Sur d’autres sites (5298)
-
WebRTC : unsync audio after processing using ffmpeg
22 novembre 2013, par QuickSilverI am recording a video and using RecordRTC : WebRTC . After receiving the webm video and wav audio at server, I'm encoding it to a mp4 file using ffmpeg(executing shell command via php). But after encoding process, the audio is unsync with video (audio ends before video). How can I fix this ?
I have noticed that the recorded audio is 1 sec less in length with video.
js code is here
record.onclick = function() {
record.disabled = true;
var video_constraints = {
mandatory: {
"minWidth": "320",
"minHeight": "240",
"minFrameRate": "24",
"maxWidth": "320",
"maxHeight": "240",
"maxFrameRate": "24"
},
optional: []
};
navigator.getUserMedia({
audio: true,
video: video_constraints
}, function(stream) {
preview.src = window.URL.createObjectURL(stream);
preview.play();
// var legalBufferValues = [256, 512, 1024, 2048, 4096, 8192, 16384];
// sample-rates in at least the range 22050 to 96000.
recordAudio = RecordRTC(stream, {
/* extra important, we need to set a big buffer when capturing audio and video at the same time*/
bufferSize: 16384
//sampleRate: 45000
});
recordVideo = RecordRTC(stream, {
type: 'video'
});
recordVideo.startRecording();
recordAudio.startRecording();
stop.disabled = false;
recording_flag = true;
$("#divcounter").show();
$("#second-step-title").text('Record your video');
initCountdown();
uploadStatus.video = false;
uploadStatus.audio = false;
});
};ffmpeg command used is :
ffmpeg -y -i 166890589.wav -i 166890589.webm -vcodec libx264 166890589.mp4Currently I'm adding an offset of -1 to ffmpeg, but i don't think it's right.
ffmpeg -y -itsoffset -1 -i 166890589.wav -i 166890589.webm -vcodec libx264 166890589.mp4 -
FFmpeg to RTMP - no audio on output [closed]
25 mars 2022, par John Mergene ArellanoFrom my client side, I am sending a stream using the Socket.IO library. I captured the video and audio using getUserMedia API.


navigator.mediaDevices.getUserMedia(constraints).then((stream) => {
 window.videoStream = video.srcObject = stream;
 let mediaRecorder = new MediaRecorder(stream, {
 videoBitsPerSecond : 3 * 1024 * 1024
 });
 mediaRecorder.addEventListener('dataavailable', (e) => {
 let data = e.data;
 socket.emit('live', data);
 });
 mediaRecorder.start(1000);
});


Then my server will receive the stream and write it to FFmpeg.


client.on('live', (stream)=>{
 if(ffmpeg)
 ffmpeg.stdin.write(stream);
});


I tried watching the live video in VLC media player. There is a 5 seconds delay and no audio output.


Please see below for FFmpeg options I used :


ffmpeg = this.CHILD_PROCESS.spawn("ffmpeg", [
 '-f',
 'lavfi',
 '-i', 'anullsrc',
 '-i','-',
 '-c:v', 'libx264', '-preset', 'veryfast', '-tune', 'zerolatency',
 '-c:a', 'aac', '-ar', '44100', '-b:a', '64k',
 '-y', //force to overwrite
 '-use_wallclock_as_timestamps', '1', // used for audio sync
 '-async', '1', // used for audio sync
 '-bufsize', '1000',
 '-f',
 'flv',
 `rtmp://127.0.0.1:1935/live/stream` ]);


What is wrong with my setup ?


I tried removing some of the options but failed. I am expecting to have an output of video and audio from the getUserMedia API.


-
Try to concatenate videos using FFmpeg Package. My videos concatenate correctly but those that i record from fornt camera rotate 90' in concatenate
24 avril 2024, par Ahmad AkramHere is my code where I pass a list of image paths that concatenate. I am facing an issue with the front camera video. When concatenated completely some videos rotate 90 degrees.


Future<void> mergeVideos(List<string> videoPaths) async {
 VideoHelper.showInSnackBar('Videos merged Start', context);
 String outputPath = await VideoHelper.generateOutputPath();
 FlutterFFmpeg flutterFFmpeg = FlutterFFmpeg();

 // Create a text file containing the paths of the videos to concatenate
 String fileListPath =
 '${(await getTemporaryDirectory()).path}/fileList.txt';
 File fileList = File(fileListPath);
 await fileList
 .writeAsString(videoPaths.map((path) => 'file \'$path\'').join('\n'));

 // Run FFmpeg command to concatenate videos
 // String command = '-f concat -safe 0 -i $fileListPath -c copy $outputPath';

 String command =
 '-f concat -safe 0 -i $fileListPath -vf "transpose=1" -c:a copy $outputPath';

 VideoHelper.showInSnackBar('command Start', context);
 await flutterFFmpeg.execute(command).then((value) {
 if (value == 0) {
 print("Output Path : $outputPath");
 VideoHelper.showInSnackBar('Videos merged successfully', context);
 Navigator.push(
 context,
 MaterialPageRoute(
 builder: (context) => VideoPlayerScreen(
 videoFile: XFile(outputPath),
 )));
 } else {
 VideoHelper.showInSnackBar(
 'Error merging videos ::::: returnCode=== $value ', context);
 }
 });
 }
</string></void>

Syndiquer tout le site
