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• ffmeg, missing size : Could not find codec parameters for stream 0 (Video : exr, gbrapf32le)

  13 septembre 2022, par RobertSmith

  I’m getting an error message using ffmpeg on Windows (a newer version of ffmpeg from 2022) using exr files. I'm trying to convert the exr to a video. However, I can get the error with just the command below :

  


  ffmpeg.exe -analyzeduration 9223372036854775807 -probesize 9223372036854775807 -i test_501_030_075_cg_testing_v001.1001.exr


  


  Below is the error :

  


  [exr @ 0000026319944c00] Unsupported channel VRayVelocity.Y.
[exr @ 0000026319944c00] Unsupported channel VRayVelocity.Z.
[exr @ 0000026319944c00] Multiple channels with index 2.
[exr @ 0000026319944c00] Wrong or missing size information.
[exr_pipe @ 0000026319932ac0] Could not find codec parameters for stream 0 (Video: exr, gbrapf32le): unspecified size
Consider increasing the value for the 'analyzeduration' (9223372036854775807) and 'probesize' (9223372036854775807) options
Input #0, exr_pipe, from 'X: Test_501_030_075_cg_testing_v001.1001.exr':
  Duration: N/A, bitrate: N/A
  Stream #0:0: Video: exr, gbrapf32le, 25 fps, 25 tbr, 25 tbn


  


  When converting to a video, I also got :

  


  2022-09-12 10:39:36:  0: [buffer @ 000001402775ee80] Unable to parse option value "0x0" as image size
2022-09-12 10:39:36:  0: [buffer @ 000001402775ee80] Error setting option video_size to value 0x0.


  


  I'm not sure what's wrong ? I can open the EXR itself. We've also used ffmpeg fine on a bunch of other exrs (which also got those errors about unsupported channels like VRayVelocity, but ffmpeg worked just fine).
I believe I have set analyzeduration and probsize to the highest values allowed ( MaxInt64).

  


  Anyone have any ideas/suggestions ? Thanks !

  


  P.S. If people need more info here, please let me know what info specifically might be helpful. I have done research on this question (setting analyzeduration to highest setting for example) I'm posting here because I'm lost as what to try next and wondering if people have had similar issues

  


• NodeJS server that converts m4a audio blobs into mp3 files works locally, but not on server

  14 juillet 2023, par Joris508

  I'm experiencing issues when running my NodeJS server on a remote host. The server is supposed to convert webm and m4a audio files to mp3 files using the ffmpeg-installer and fluent-ffmpeg (npm) packages.

  


  The original files are created using the MediaRecorder Web API. Chrome supports the 'audio/webm' (webm) mimeType, but Safari only supports 'audio/mp4' (m4a). When I run my node server locally, it converts both the m4a files and the webm files just fine. When I host my node server, the webm files are converted without problems, but the m4a files specifically return as 1 second audio files. They're all 8398 bytes in size. You can hear the start of the recording in the 1 second file as well. I'm completely lost on what the issue could be, especially since it works fine when the node server runs locally (through the terminal).

  


  Does anyone have any ideas ? I checked both the node version as well as all the packages ; they are identical. The server doesn't give any errors or feedback. It behaves as if everything is going fine.

  


  I simplified the code and selected just the part that converts the file. Here's the code that converts the m4a file to an mp3. It's triggered 2 seconds after the server starts up (the file is already there on startup in this testing scenario) :

  


  const convertFunc = () => {
    ffmpeg('./public/testfile.m4a')
        .output('./public/testfile.mp3')
        .on('error', (err) => {
            console.error('Error: ' + err.message);
        })
        .on('end', () => {
            console.log('Conversion completed successfully');
        })
        .run()
}


  


• Re-stream (forward) UDP-Live-Video-Stream (using winsock) reduce video quality ?

  18 août 2014, par vantrung -cuncon

  So, I used ffmpeg to stream the live webcam using UDP protocol to port 1111 :

  ffmpeg -f dshow -i video="Lenovo EasyCamera" -f mpegts udp://localhost:1111

  When I played it directly by ffplay from port 1111, everything worked properly :

  ffplay udp://localhost:1111

  I got the video quality like this :

  

  So I think, I could write some winsock-codes to listen to port 1111 & forward any UDP packet that it catches to port 2222. Thus, I could simulate that I’m streaming to port 2222. My code is something like this :

  ' // Please note that this is the simplified code - cause it worked
  
  ' // i've just post the key lines
  
  Winsock1.Bind 1111
  
  Winsock2.remotePort = 2222
  
  
  
  WinSock1.GetData myPacket
  
  Winsock2.SendData myPacket

  Then I tried playing the stream from port 2222 using ffplay :

  ffplay udp://localhost:2222

  Well, I don’t know why the video quality turned to this bad :

  

  The point is that, I’ve sent the same UDP packets in the same order as the streaming-source. What could be wrong here ?

  ---

  PS : I’ve tried a similar experiment like above with TCP, but the end-result video quality was as good as direct streaming. So, could this be a problem of UDP ?

  ---

  PS2 : I’ve tested the UDP packet loss & disorder by replacing the ffplay with a socket that listen to port 2222 & print out all the received packets. But the result is all 10,000+ packets were in correct order & nothing lost. What a crazy phenomenon ?