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Autres articles (14)
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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 -
Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
Les vidéos
21 avril 2011, parComme les documents de type "audio", Mediaspip affiche dans la mesure du possible les vidéos grâce à la balise html5 .
Un des inconvénients de cette balise est qu’elle n’est pas reconnue correctement par certains navigateurs (Internet Explorer pour ne pas le nommer) et que chaque navigateur ne gère en natif que certains formats de vidéos.
Son avantage principal quant à lui est de bénéficier de la prise en charge native de vidéos dans les navigateur et donc de se passer de l’utilisation de Flash et (...)
Sur d’autres sites (3740)
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FFmpeg error : ratecontrol_init : can't open stats file
18 décembre 2015, par oldo.nichoI’ve setup an AWS EC2 instance running Ubuntu 14.04 and have installed FFmpeg so that I can compress and transcode video.
I’m trying to do a two pass conversion with the following code :
ffmpeg -i input-file.avi -codec:v libx264 -profile:v high -preset slow -b:v 500k -maxrate 500k -bufsize 1000k -vf scale=702:-1 -threads 0 -pass 1 -an -f mp4 ~/encoded/nulland second pass :
ffmpeg -i input-file.avi -codec:v libx264 -profile:v high -preset slow -b:v 500k -maxrate 500k -bufsize 1000k -vf scale=702:-1 -threads 0 -pass 2 -codec:a libfdk_aac -b:a 128k -f mp4 output-file.mp4However I get the following error :
ffmpeg version N-77283-g91c2a33 Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04)
configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ubuntu/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --bindir=/home/ubuntu/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree
libavutil 55. 11.100 / 55. 11.100
libavcodec 57. 17.100 / 57. 17.100
libavformat 57. 20.100 / 57. 20.100
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 21.100 / 6. 21.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Input #0, avi, from 'input-file.avi':
Duration: 01:18:05.29, start: 0.000000, bitrate: 2025 kb/s
Stream #0:0: Video: mpeg4 (Simple Profile) (XVID / 0x44495658), yuv420p, 720x480 [SAR 1:1 DAR 3:2], 1789 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn, 29.97 tbc
Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp, 224 kb/s
[libx264 @ 0x1e04240] using SAR=1/1
[libx264 @ 0x1e04240] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX AVX2 FMA3 LZCNT BMI2
[libx264 @ 0x1e04240] ratecontrol_init: can't open stats file
Output #0, mp4, to '/home/ubuntu/encoded/null':
Stream #0:0: Video: h264, none, q=2-31, 128 kb/s, SAR 1:1 DAR 0:0, 29.97 fps
Metadata:
encoder : Lavc57.17.100 libx264
Stream mapping:
Stream #0:0 -> #0:0 (mpeg4 (native) -> h264 (libx264))
Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or heightThe command as written above works fine on my local computer (running OSX). Would anyone have any suggestions as to how to fix this problem ?
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avformat/iamfdec : Check side_substream_id before use
22 avril, par Michael Niedermayer -
ffmpeg pipe process ends right after writing first buffer data to input stream and does not keep running
6 mai, par Taketo MatsunagaI have been trying to convert 16bit PCM (s16le) audio data to webm using ffmpeg in C#.
But the process ends right after the writing the first buffer data to standard input.
I has exited with the status 0, meaning success. But do not know why....
Could anyone tell me why ?


I apprecite it if you could support me.


public class SpeechService : ISpeechService
 {
 
 /// <summary>
 /// Defines the _audioInputStream
 /// </summary>
 private readonly MemoryStream _audioInputStream = new MemoryStream();

 public async Task SendPcmAsWebmViaWebSocketAsync(
 MemoryStream pcmAudioStream,
 int sampleRate,
 int channels) 
 {
 string inputFormat = "s16le";

 var ffmpegProcessInfo = new ProcessStartInfo
 {
 FileName = _ffmpegPath,
 Arguments =
 $"-f {inputFormat} -ar {sampleRate} -ac {channels} -i pipe:0 " +
 $"-f webm pipe:1",
 RedirectStandardInput = true,
 RedirectStandardOutput = true,
 RedirectStandardError = true,
 UseShellExecute = false,
 CreateNoWindow = true,
 };

 _ffmpegProcess = new Process { StartInfo = ffmpegProcessInfo };

 Console.WriteLine("Starting FFmpeg process...");
 try
 {

 if (!await Task.Run(() => _ffmpegProcess.Start()))
 {
 Console.Error.WriteLine("Failed to start FFmpeg process.");
 return;
 }
 Console.WriteLine("FFmpeg process started.");

 }
 catch (Exception ex)
 {
 Console.Error.WriteLine($"Error starting FFmpeg process: {ex.Message}");
 throw;
 }

 var encodeAndSendTask = Task.Run(async () =>
 {
 try
 {
 using var ffmpegOutputStream = _ffmpegProcess.StandardOutput.BaseStream;
 byte[] buffer = new byte[8192]; // Temporary buffer to read data
 byte[] sendBuffer = new byte[8192]; // Buffer to accumulate data for sending
 int sendBufferIndex = 0; // Tracks the current size of sendBuffer
 int bytesRead;

 Console.WriteLine("Reading WebM output from FFmpeg and sending via WebSocket...");
 while (true)
 {
 if ((bytesRead = await ffmpegOutputStream.ReadAsync(buffer, 0, buffer.Length)) > 0)
 {
 // Copy data to sendBuffer
 Array.Copy(buffer, 0, sendBuffer, sendBufferIndex, bytesRead);
 sendBufferIndex += bytesRead;

 // If sendBuffer is full, send it via WebSocket
 if (sendBufferIndex >= sendBuffer.Length)
 {
 var segment = new ArraySegment<byte>(sendBuffer, 0, sendBuffer.Length);
 _ws.SendMessage(segment);
 sendBufferIndex = 0; // Reset the index after sending
 }
 }
 }
 }
 catch (OperationCanceledException)
 {
 Console.WriteLine("Encode/Send operation cancelled.");
 }
 catch (IOException ex) when (ex.InnerException is ObjectDisposedException)
 {
 Console.WriteLine("Stream was closed, likely due to process exit or cancellation.");
 }
 catch (Exception ex)
 {
 Console.Error.WriteLine($"Error during encoding/sending: {ex}");
 }
 });

 var errorReadTask = Task.Run(async () =>
 {
 Console.WriteLine("Starting to read FFmpeg stderr...");
 using var errorReader = _ffmpegProcess.StandardError;
 try
 {
 string? line;
 while ((line = await errorReader.ReadLineAsync()) != null) 
 {
 Console.WriteLine($"[FFmpeg stderr] {line}");
 }
 }
 catch (OperationCanceledException) { Console.WriteLine("FFmpeg stderr reading cancelled."); }
 catch (TimeoutException) { Console.WriteLine("FFmpeg stderr reading timed out (due to cancellation)."); }
 catch (Exception ex) { Console.Error.WriteLine($"Error reading FFmpeg stderr: {ex.Message}"); }
 Console.WriteLine("Finished reading FFmpeg stderr.");
 });

 }

 public async Task AppendAudioBuffer(AudioMediaBuffer audioBuffer)
 {
 try
 {
 // audio for a 1:1 call
 var bufferLength = audioBuffer.Length;
 if (bufferLength > 0)
 {
 var buffer = new byte[bufferLength];
 Marshal.Copy(audioBuffer.Data, buffer, 0, (int)bufferLength);

 _logger.Info("_ffmpegProcess.HasExited:" + _ffmpegProcess.HasExited);
 using var ffmpegInputStream = _ffmpegProcess.StandardInput.BaseStream;
 await ffmpegInputStream.WriteAsync(buffer, 0, buffer.Length);
 await ffmpegInputStream.FlushAsync(); // バッファをフラッシュ
 _logger.Info("Wrote buffer data.");

 }
 }
 catch (Exception e)
 {
 _logger.Error(e, "Exception happend writing to input stream");
 }
 }

</byte>

Starting FFmpeg process...
FFmpeg process started.
Starting to read FFmpeg stderr...
Reading WebM output from FFmpeg and sending via WebSocket...
[FFmpeg stderr] ffmpeg version 7.1.1-essentials_build-www.gyan.dev Copyright (c) 2000-2025 the FFmpeg developers
[FFmpeg stderr] built with gcc 14.2.0 (Rev1, Built by MSYS2 project)
[FFmpeg stderr] configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
[FFmpeg stderr] libavutil 59. 39.100 / 59. 39.100
[FFmpeg stderr] libavcodec 61. 19.101 / 61. 19.101
[FFmpeg stderr] libavformat 61. 7.100 / 61. 7.100
[FFmpeg stderr] libavdevice 61. 3.100 / 61. 3.100
[FFmpeg stderr] libavfilter 10. 4.100 / 10. 4.100
[FFmpeg stderr] libswscale 8. 3.100 / 8. 3.100
[FFmpeg stderr] libswresample 5. 3.100 / 5. 3.100
[FFmpeg stderr] libpostproc 58. 3.100 / 58. 3.100

[2025-05-06 15:44:43,598][INFO][XbLogger.cs:85] _ffmpegProcess.HasExited:False
[2025-05-06 15:44:43,613][INFO][XbLogger.cs:85] Wrote buffer data.
[2025-05-06 15:44:43,613][INFO][XbLogger.cs:85] Wrote buffer data.
[FFmpeg stderr] [aist#0:0/pcm_s16le @ 0000025ec8d36040] Guessed Channel Layout: mono
[FFmpeg stderr] Input #0, s16le, from 'pipe:0':
[FFmpeg stderr] Duration: N/A, bitrate: 256 kb/s
[FFmpeg stderr] Stream #0:0: Audio: pcm_s16le, 16000 Hz, mono, s16, 256 kb/s
[FFmpeg stderr] Stream mapping:
[FFmpeg stderr] Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))
[FFmpeg stderr] [libopus @ 0000025ec8d317c0] No bit rate set. Defaulting to 64000 bps.
[FFmpeg stderr] Output #0, webm, to 'pipe:1':
[FFmpeg stderr] Metadata:
[FFmpeg stderr] encoder : Lavf61.7.100
[FFmpeg stderr] Stream #0:0: Audio: opus, 16000 Hz, mono, s16, 64 kb/s
[FFmpeg stderr] Metadata:
[FFmpeg stderr] encoder : Lavc61.19.101 libopus
[FFmpeg stderr] [out#0/webm @ 0000025ec8d36200] video:0KiB audio:1KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 67.493113%
[FFmpeg stderr] size= 1KiB time=00:00:00.04 bitrate= 243.2kbits/s speed=2.81x
Finished reading FFmpeg stderr.
[2025-05-06 15:44:44,101][INFO][XbLogger.cs:85] _ffmpegProcess.HasExited:True
[2025-05-06 15:44:44,132][ERROR][XbLogger.cs:67] Exception happend writing to input stream
System.ObjectDisposedException: Cannot access a closed file.
 at System.IO.FileStream.WriteAsync(Byte[] buffer, Int32 offset, Int32 count, CancellationToken cancellationToken)
 at System.IO.Stream.WriteAsync(Byte[] buffer, Int32 offset, Int32 count)
 at EchoBot.Media.SpeechService.AppendAudioBuffer(AudioMediaBuffer audioBuffer) in C:\Users\tm068\Documents\workspace\myprj\xbridge-teams-bot\src\EchoBot\Media\SpeechService.cs:line 242


I am expecting the ffmpeg process keep running.


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