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Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
Autres articles (28)
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Submit bugs and patches
13 avril 2011Unfortunately a software is never perfect.
If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
You may also (...) -
MediaSPIP Player : les contrôles
26 mai 2010, parLes contrôles à la souris du lecteur
En plus des actions au click sur les boutons visibles de l’interface du lecteur, il est également possible d’effectuer d’autres actions grâce à la souris : Click : en cliquant sur la vidéo ou sur le logo du son, celui ci se mettra en lecture ou en pause en fonction de son état actuel ; Molette (roulement) : en plaçant la souris sur l’espace utilisé par le média (hover), la molette de la souris n’exerce plus l’effet habituel de scroll de la page, mais diminue ou (...) -
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 (4315)
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ffmpeg rtp-stream with gsm-codec
15 octobre 2020, par BirgitI want to use ffmpeg for encoding and decoding gsm. I built ffmpeg with the
--enable-libgsm
option.

I can now use the ffmpeg-command-line-tool to read gsm-encoded files, convert files to gsm, and also receive a gsm-encoded rtp stream.
So therefore I think the gsm-encoder and gsm-decoder are working properly.


But for some reason I am not able to send and gsm-encoded rtp-stream.


I tried the following comands :


ffmpeg -re -i test.wav -c:a libgsm -f rtp rtp://127.0.0.1:5000


ffmpeg -re -i test.wav -c:a gsm -f rtp rtp://127.0.0.1:5000


I receive the error :
Unsupported codec gsm. Could not write header for output file.


I tried to use gdb to see what's going on. I think the problem is that in the file
libavformat/rtpenc.c:49
gsm is not under the supported codecs. Does that mean it is not possible to use ffmpeg to create a gsm-encoded rtp-stream ? Is there a workaround, to overcome this issue ?

I would appreciate any help and hints what I could try. :)


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how to send the input data to FFMPEG from a C# program
18 octobre 2020, par jstuardoI need to send a binary stream to FFMPEG so that it sends to an RTMP server.


I did it in a nodejs script using socket.io library and in Linux. It works perfectly.


I need to do the same, but in a Windows Forms application using C#.


This is how I run the ffmpeg.exe application :


_currentProcess = new Process();
 _currentProcess.StartInfo.FileName = _ffmpegExe;
 _currentProcess.StartInfo.Arguments = BuildOptions(framesPerSecond, audioBitRate, audioEncoding, rtmpServer);
 _currentProcess.StartInfo.UseShellExecute = false;
 _currentProcess.StartInfo.CreateNoWindow = true;
 _currentProcess.StartInfo.RedirectStandardInput = true;
 _currentProcess.StartInfo.RedirectStandardError = true;
 _currentProcess.ErrorDataReceived += CurrentProcess_ErrorDataReceived;
 _currentProcess.Start();
 _currentProcess.BeginErrorReadLine();



BuildOptions
method is defined this way :

private string BuildOptions(int framesPerSecond, int audioBitRate, string audioEncoding, string rtmpServer)
 {
 string options;
 if (framesPerSecond == 1)
 {
 options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency -r 1 -g 2 -keyint_min 2 -x264opts keyint=2 -crf 25 -pix_fmt yuv420p -profile:v baseline -level 3 -c:a aac -b:a {audioEncoding} -ar {audioBitRate}-f flv {rtmpServer}";
 }
 else if (framesPerSecond == 15)
 {
 options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency max_muxing_queue_size 1000 -bufsize 5000 -r 15 -g 30 -keyint_min 30 -x264opts keyint=30 -crf 25 -pix_fmt yuv420p -profile:v baseline -level 3 -c:a aac -b:a {audioEncoding} -ar {audioBitRate} -f flv {rtmpServer}";
 }
 else
 {
 options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency -c:a aac -ar {audioBitRate} -b:a {audioEncoding} -bufsize 5000 -f flv {rtmpServer}";
 }

 return options;
 }



I am sending the data to the standard input this way :


public void EncodeAndSend(byte[] data)
 {
 if (_currentProcess != null)
 {
 var streamWriter = _currentProcess.StandardInput;
 streamWriter.Write(Encoding.GetEncoding("ISO-8859-1").GetChars(data));
 }
 }



And finally, this method is for receiving the standard error which receives the result from ffmpeg.exe :


private void CurrentProcess_ErrorDataReceived(object sender, DataReceivedEventArgs e)
 {
 Console.WriteLine(e.Data);
 }



When I run the application, this is shown in the console :


ffmpeg version 4.3.1-2020-10-01-essentials_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 10.2.0 (Rev3, Built by MSYS2 project)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --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-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --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
 libavutil 56. 51.100 / 56. 51.100
 libavcodec 58. 91.100 / 58. 91.100
 libavformat 58. 45.100 / 58. 45.100
 libavdevice 58. 10.100 / 58. 10.100
 libavfilter 7. 85.100 / 7. 85.100
 libswscale 5. 7.100 / 5. 7.100
 libswresample 3. 7.100 / 3. 7.100
 libpostproc 55. 7.100 / 55. 7.100
pipe:: Invalid data found when processing input



If I change the
EncodeAndSend
method to be :

public void EncodeAndSend(byte[] data)
 {
 if (_currentProcess != null)
 {
 var streamWriter = _currentProcess.StandardInput;
 streamWriter.Write(data);
 }
 }



pipe:: Invalid data found when processing input
error is not produced, but no more outputs are shown so it seems it is not working.

What is wrong with this ? how can I send the data to the FFMPEG process ?


Finally, I tell you that the binary stream comes from the camera by mean of MediaRecorder in a web page (the same used for my program in nodejs server, so that it is not the issue here)


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How to restream multicast stream with ffmpeg
26 octobre 2020, par verbI am new to ffmpeg and need to restream multicast and scale it. Tried different parameters and i have managed to restream and scale but it always appear some pat,pmt or pcr error and som interuptions in the stream appear.The input stream is cbr 14Mbit and i try to set the bitrate as 6Mbit please check my config and if you notice something wrong let me know :




ffmpeg -re -i "udp ://@238.252.250.9:5000 ?overrun_nonfatal=1&fifo_size=1000000&bitrate=70000000&pkt_size=188" -map 0:0 -map 0:2 -b:v 3000k -minrate 3000k -maxrate 4000k -bufsize 8000K -pcr_period 20 -flush_packets 0 -tune zerolatency -preset ultrafast -threads 2 -c:a copy -qmax 12 -f mpegts -muxrate 6M "udp ://@239.253.251.13:5505 ?pkt_size=188&overrun_nonfatal=1&localaddr=10.253.251.66&bitrate=6000000"




here is the input stream :


Input #0, mpegts, from 'udp://@238.252.250.9:5000':
 Duration: N/A, start: 46612.831967, bitrate: N/A
 Program 2002 
 Metadata:
 service_name : RT Doc HD
 service_provider: GLOBECAST
 Stream #0:0[0x7e5]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc
 Stream #0:1[0x7e6](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s
 Stream #0:2[0x7e7](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 192 kb/s



I don't understand all parameters especially the parameters concerning input/output udp stream so please help me to solve the correct command.