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Médias (1)
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GetID3 - Bloc informations de fichiers
9 avril 2013, par
Mis à jour : Mai 2013
Langue : français
Type : Image
Autres articles (37)
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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) (...)
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L’utiliser, en parler, le critiquer
10 avril 2011La première attitude à adopter est d’en parler, soit directement avec les personnes impliquées dans son développement, soit autour de vous pour convaincre de nouvelles personnes à l’utiliser.
Plus la communauté sera nombreuse et plus les évolutions seront rapides ...
Une liste de discussion est disponible pour tout échange entre utilisateurs.
Sur d’autres sites (6727)
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Could not find ffmpeg executable, tried "/srv/linux-x64/ffmpeg"
20 mai 2024, par AlanI am trying to run my docker image on AWS EC2 Instance. However, I am having a trouble with the image file. When I run a task in AWS ECS, it says :


Could not find ffmpeg executable, tried "/srv/linux-x64/ffmpeg", "/srv/app/node_modules/@ffmpeg-installer/linux-x64/ffmpeg" and "/srv/app/node_modules/@ffmpeg-installer/linux-x64/ffmpeg"



This is my Dockerfile :


FROM node:latest as server
WORKDIR /srv/app
COPY ./server/package*.json ./
RUN apt-get update && apt-get install -y 'ffmpeg'
RUN npm install
COPY ./server .
RUN npm run build

FROM node:latest as client
WORKDIR /srv/app
COPY ./client/package*.json ./
RUN npm install
COPY ./client .
RUN npm run build

FROM node:latest as production
WORKDIR /srv/app
RUN mkdir /public
COPY --from=server /srv/app/dist ./
COPY --from=client /srv/app/dist ./public
ENV NODE_ENV=production
EXPOSE 5000
CMD ["node", "app.js"]



This is my package.json :


},
 "keywords": [],
 "author": "",
 "license": "ISC",
 "dependencies": {
 "@ffmpeg-installer/ffmpeg": "^1.1.0",
 "@types/ffmpeg": "^1.0.7",
 "@types/fluent-ffmpeg": "^2.1.24"
 }



This is how I used ffmpeg in my code :


import ffmpegPath from '@ffmpeg-installer/ffmpeg';
import ffmpeg from 'fluent-ffmpeg';
ffmpeg.setFfmpegPath(ffmpegPath.path);



Please help me fix this error.


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C# server problem with FFmpeg Visual Studio 2022
16 mai 2024, par seanofdeadHello everyone I've spent countless hours on my server I'm trying to create that will share video from one stream to another client computer. That is the goal anyway. I'm using Visual Studio 2022 most recent update. Currently this is my code


using System;
using System.IO;
using System.Net;
using System.Net.Sockets;
using System.Threading.Tasks;
using FFmpeg.AutoGen;
using NLog;

public class VideoStreamServer
{
 private const int Port = 5000;
 private TcpListener server;
 private FFmpegVideoEncoder encoder;
 private static readonly Logger logger = LogManager.GetCurrentClassLogger();

 public VideoStreamServer()
 {
 // Initialize FFmpeg network components
 Console.WriteLine("Initializing FFmpeg network components...");
 int result = ffmpeg.avformat_network_init();
 if (result != 0)
 {
 throw new ApplicationException($"Failed to initialize FFmpeg network components: {result}");
 }

 // Initialize encoder
 encoder = new FFmpegVideoEncoder(1920, 1080, AVCodecID.AV_CODEC_ID_H264); // Screen size & codec
 }

 public static void Main()
 {
 // Initialize NLog
 var logger = LogManager.GetCurrentClassLogger();
 logger.Info("Application started");

 var server = new VideoStreamServer();
 server.Start().Wait();
 }

 public async Task Start()
 {
 server = new TcpListener(IPAddress.Any, Port);
 server.Start();
 logger.Info($"Server started on port {Port}...");

 try
 {
 while (true)
 {
 TcpClient client = await server.AcceptTcpClientAsync();
 logger.Info("Client connected...");
 _ = HandleClientAsync(client);
 }
 }
 catch (Exception ex)
 {
 logger.Error(ex, "An error occurred");
 }
 }

 private async Task HandleClientAsync(TcpClient client)
 {
 using (client)
 using (NetworkStream netStream = client.GetStream())
 {
 try
 {
 await encoder.StreamEncodedVideoAsync(netStream);
 }
 catch (IOException ex)
 {
 logger.Error(ex, $"Network error: {ex.Message}");
 // Additional logging and recovery actions
 }
 catch (Exception ex)
 {
 logger.Error(ex, $"Unexpected error: {ex.Message}");
 // Log and handle other exceptions
 }
 finally
 {
 // Ensure all resources are cleaned up properly
 client.Close();
 logger.Info("Client connection closed properly.");
 }
 }
 }
}

public class FFmpegVideoEncoder
{
 private readonly int width;
 private readonly int height;
 private readonly AVCodecID codecId;
 private static readonly Logger logger = LogManager.GetCurrentClassLogger();

 public FFmpegVideoEncoder(int width, int height, AVCodecID codecId)
 {
 this.width = width;
 this.height = height;
 this.codecId = codecId;
 InitializeEncoder();
 }

 private void InitializeEncoder()
 {
 // Initialize FFmpeg encoder here
 logger.Debug("FFmpeg encoder initialized");
 }

 public async Task StreamEncodedVideoAsync(NetworkStream netStream)
 {
 // Initialize reusable buffers for efficiency
 byte[] buffer = new byte[4096];

 // Capture and encode video frames
 while (true)
 {
 try
 {
 // Simulate capture and encoding
 byte[] videoData = new byte[1024]; // This would come from the encoder

 // Simulate streaming
 for (int i = 0; i < videoData.Length; i += buffer.Length)
 {
 int chunkSize = Math.Min(buffer.Length, videoData.Length - i);
 Buffer.BlockCopy(videoData, i, buffer, 0, chunkSize);
 await netStream.WriteAsync(buffer, 0, chunkSize);
 }
 }
 catch (Exception ex)
 {
 logger.Error(ex, "Error streaming video data");
 throw; // Rethrow to handle in the calling function
 }

 await Task.Delay(1000 / 30); // For 30 FPS
 }
 }
}



When I build no issues but when i run it in debug mode I get this error


System.DllNotFoundException
 HResult=0x80131524
 Message=Unable to load DLL 'avformat.59 under C:\Users\phlfo\Downloads\cfolder\server\ConsoleApp\bin\Debug\': The specified module could not be found.
 Source=FFmpeg.AutoGen
 StackTrace:
 at FFmpeg.AutoGen.ffmpeg.LoadLibrary(String libraryName, Boolean throwException)
 at FFmpeg.AutoGen.ffmpeg.<>c.<.cctor>b__7_0(String libraryName)
 at FFmpeg.AutoGen.ffmpeg.<>c.<.cctor>b__7_242()
 at FFmpeg.AutoGen.ffmpeg.avformat_network_init()
 at VideoStreamServer..ctor() in C:\Users\phlfo\Downloads\cfolder\server\ConsoleApp\Program.cs:line 20
 at VideoStreamServer.Main() in C:\Users\phlfo\Downloads\cfolder\server\ConsoleApp\Program.cs:line 36



I have FFmpeg.AutoGen package installed through VS i originally started with 7.0 but then switched to 5.0 to see if an older version might work (it did not). I also have the files in the same directory as the .exe for testing purposes as seen in the screenshot. Can someone please help me figure out this error.



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Why every audio part is louder in FFmpeg when I join them in one audio ?
14 mai 2024, par Volodymyr BilovusI trying to make dubbing for audio. I have original audio track and I want to put translated audio parts on top of the original.


translated audio 100% vol : —p1--- ---p2— -----p3--- —p4—


original audio 5% vol : -----------------------------------------


Here is my FFmpeg command with filter_complex


ffmpeg -i video_wpmXlZF4XiE.opus -i 989-audio.mp3 -i 989-audio.mp3 -i 989-audio.mp3 -i 989-audio.mp3 \
-filter_complex "\
[0:a]loudnorm=I=-14:TP=-2:LRA=7, volume=0.05[original]; \
[1:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=5000|5000, volume=1.0[sent1]; \
[2:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=10000|10000, volume=1.0[sent2]; \
[3:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=20000|20000, volume=1.0[sent3]; \
[4:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=30000|30000, volume=1.0[sent4]; \
[original][sent1][sent2][sent3][sent4]amix=inputs=5:duration=longest[out]" \
-map "[out]" output.mp3



Audios I put on top of the original audio track is the same
-i 989-audio.mp3
I made it by purpose to show the problem
And here is the audio levels on final generated track.


As you can see, first and second only slightly different but third
and fourth have totally different(higher) volume level (Notice, audio is the same).
Why it's happened ? And how can I workaround this odd behaviour ?