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999,999
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Demon seed (wav version)
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Mis à jour : Avril 2013
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Mis à jour : Avril 2013
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Mis à jour : Avril 2013
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Mis à jour : Avril 2013
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Type : Audio
Autres articles (66)
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A discussion list is available for all exchanges between users. -
Gestion des droits de création et d’édition des objets
8 février 2011, parPar défaut, beaucoup de fonctionnalités sont limitées aux administrateurs mais restent configurables indépendamment pour modifier leur statut minimal d’utilisation notamment : la rédaction de contenus sur le site modifiables dans la gestion des templates de formulaires ; l’ajout de notes aux articles ; l’ajout de légendes et d’annotations sur les images ;
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Sur d’autres sites (11326)
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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)


-
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. :)


-
Python vlc stream from outside NAT
7 octobre 2020, par xKedarI'm trying to stream, using FFmpeg, my webcam and audio from PC1 to PC2 in another LAN.


PC1 : Public IP address with port forwarding so I can reach it


PC2 : In a different NAT from PC1


I basically run a server on PC1 in order to acquire IP and port from PC2 and reply on the same address


import socket

 localPort = 1234
 bufferSize = 1024

 UDPServerSocket = socket.socket(family=socket.AF_INET, type=socket.SOCK_DGRAM)
 UDPServerSocket.bind(("", localPort)) # Bind to address and port

 while(True):
 bytesAddressPair = UDPServerSocket.recvfrom(bufferSize)
 message = bytesAddressPair[0].decode("utf-8")
 address = bytesAddressPair[1]
 # Sending a reply to client
 UDPServerSocket.sendto(str.encode("Hello"), address)
 break

 UDPServerSocket.close()



Then I try to send the stream with the same port number both for the server(localPort) and the client(the one I acquired from address)


import re
 from threading import Thread
 from subprocess import Popen, PIPE

 def detect_devices():
 list_cmd = 'ffmpeg -list_devices true -f dshow -i dummy'.split()
 p = Popen(list_cmd, stderr=PIPE)
 flagcam = flagmic = False
 for line in iter(p.stderr.readline,''):
 if flagcam:
 cam = re.search('".*"',line.decode(encoding='UTF-8')).group(0)
 cam = cam if cam else ''
 flagcam = False
 if flagmic:
 mic = re.search('".*"',line.decode(encoding='UTF-8')).group(0)
 mic = mic if mic else ''
 flagmic = False
 elif 'DirectShow video devices'.encode(encoding='UTF-8') in line:
 flagcam = True
 elif 'DirectShow audio devices'.encode(encoding='UTF-8') in line:
 flagmic = True
 elif 'Immediate exit requested'.encode(encoding='UTF-8') in line:
 break
 return cam, mic 


 class ffmpegThread (Thread):
 def __init__(self, address):
 Thread.__init__(self)
 self.address = address

 def run(self):
 cam, mic = detect_devices()
 command = 'ffmpeg -f dshow -i video='+cam+':audio='+mic+' -profile:v high -pix_fmt yuvj420p -level:v 4.1 -preset ultrafast -tune zerolatency -vcodec libx264 -r 10 -b:v 512k -s 240x160 -acodec aac -ac 2 -ab 32k -ar 44100 -f mpegts -flush_packets 0 -t 40 udp://'+self.address+'?pkt_size=1316?localport='+str(localPort)
 p = Popen(command , stderr=PIPE)
 for line in iter(p.stderr.readline,''):
 if len(line) <5: break
 p.terminate()

 thread1 = ffmpegThread(address[0]+":"+str(address[1]))
 thread1.start()



While on the other side(PC2) I have :


from threading import Thread
 import tkinter as tk
 import vlc

 class myframe(tk.Frame):
 def __init__(self, width=240, height=160):
 self.root = tk.Tk()
 super(myframe, self).__init__(self.root)
 self.root.geometry("%dx%d" % (width, height))
 self.root.wm_attributes("-topmost", 1)
 self.grid()
 self.frame = tk.Frame(self, width=240, height=160)
 self.frame.configure(bg="black")
 self.frame.grid(row=0, column=0, columnspan=2)
 self.play()
 self.root.mainloop()

 def play(self):
 self.player = vlc.Instance().media_player_new()
 self.player.set_mrl('udp://@0.0.0.0:5000')
 self.player.set_hwnd(self.frame.winfo_id())
 self.player.play()

 class guiThread (Thread):
 def __init__(self, nome):
 Thread.__init__(self)
 self.nome = nome

 def run(self):
 app = myframe()



and :


import socket

 msgFromClient = "Hello UDP Server"
 bytesToSend = str.encode(msgFromClient)
 serverAddressPort = ("MYglobal_IPaddress", 1234)
 bufferSize = 1024
 localPort = 5000

 # Create a UDP socket at client side
 UDPClientSocket = socket.socket(family=socket.AF_INET, type=socket.SOCK_DGRAM) 
 UDPClientSocket.bind(("", localPort))

 UDPClientSocket.sendto(bytesToSend, serverAddressPort)

 msgFromServer = UDPClientSocket.recvfrom(bufferSize)
 msg = msgFromServer[0].decode("utf-8")
 print(msg)
 UDPClientSocket.close()
 gui = guiThread("ThreadGUI")
 gui.start()



Where I basically try to reach the server both to send my IP:Port and to punch a hole in the NAT in order to be able to get the packages sent from PC1 despite being behind a NAT.


I think it is not working because I can not reach PC2 but I really can not figure out how to fix that because I was expecting that the first part, where I reach PC1 from PC2 was enough in order to establish a connection