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• lavf : Use wchar functions for filenames on windows for mkdir/rmdir/rename/unlink

  17 novembre 2014, par Martin Storsjö

  lavf : Use wchar functions for filenames on windows for mkdir/rmdir/rename/unlink
  This makes sure that the internal utf8 path names are handled
  
  properly - the normal file handling functions assume path names
  
  are in the native codepage, which isn’t utf8.
  This assumes that the tools outside of lavf don’t use the mkdir
  
  definition. (The tools don’t do the same reading of command line
  
  parameters as wchar either - they probably won’t handle all possible
  
  unicode file parameters properly, but at least work more predictably
  
  if no utf8/wchar conversion is involved.)
  This is moved further down in os_support.h, since windows.h shouldn’t
  
  be included before winsock2.h, while io.h needs to be included before
  
  the manual defines for lseek functions.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] libavformat/internal.h
  • [DBH] libavformat/os_support.h

• Getting garbled frames when exporting video to image sequence in FFMPEG

  28 janvier 2018, par film_guy01

  I’m very new to FFMPEG to go easy on me.

  So I’ve just started playing around with ffmpeg but I’m getting some promising results. But I keep getting images like this : https://i.imgur.com/wIMkVoV.jpg which is not what that image it supposed to look like (in case you were uncertain).

  here’s been my command line code :

  ffmpeg -i inputfile.mkv -q:v 1 -r 0.20 outputfile_%04d.jpg

  My input file is a .h264 compressed ultra high definition video saved as .mkv format.

  I’ll get error messages like this :

  [h264 @ 0x7fbccd84ce00] co located POCs unavailable
  
  [h264 @ 0x7fbccd81e200] error while decoding MB 45 61, bytestream -17 drop=406 speed=1.71x    
  
  [h264 @ 0x7fbccd81e200] concealing 17764 DC, 17764 AC, 17764 MV errors in P frame
  
  [h264 @ 0x7fbccd84ce00] co located POCs unavailable
  
  [h264 @ 0x7fbccd814400] reference picture missing during reorder
  
  [h264 @ 0x7fbccd814400] Missing reference picture, default is 65986
  
  [h264 @ 0x7fbccd814400] error while decoding MB 71 100, bytestream -26drop=598 speed=1.79x    
  
  [h264 @ 0x7fbccd814400] concealing 8378 DC, 8378 AC, 8378 MV errors in B frame
  
  [h264 @ 0x7fbccd81dc00] co located POCs unavailable
  
  [h264 @ 0x7fbccd81e200] co located POCs unavailable
  
  [h264 @ 0x7fbccd863200] co located POCs unavailable
  
  [h264 @ 0x7fbccd814400] error while decoding MB 43 116, bytestream -5 drop=710 speed=1.81x   

  Sometimes I get these errors as well, but I’m not sure they’re related.

  [h264 @ 0x7fbcce004c00] Stream #0: not enough frames to estimate rate; consider increasing probesize
  
  [swscaler @ 0x7fbccf0c8000] deprecated pixel format used, make sure you did set range correctly

  Any idea how to fix this problem and get ffmpeg to export the images cleanly ? This would be HUGE if I could figure out how to get this working.

  Thanks !

• Real time audio streaming from ffmpeg to browser (am I missing something ?)

  19 septembre 2022, par Яктенс Тид

  I have tried a couple of solutions already, but nothing works for me.
I want to stream audio from my PC to another computer with almost zero latency. Things are working fine so far in a sense of lagging and everything, sound is clear and not choppy at all, but there is something like a delay between the moment when audio starts playing on my PC and remote PC. For example when I click on Youtube 'play' button audio starts playing only after 3-4 seconds on the remote machine. The same when I click 'Pause', the sound on the remote PC stops after a couple of seconds.

  


  I've tried to use websockets\plain audio tag, but no luck so far.

  


  For example this is my solution by using websockets and pipes :

  


  def create_pipe():
    return win32pipe.CreateNamedPipe(r'\\.\pipe\__audio_ffmpeg', win32pipe.PIPE_ACCESS_INBOUND,
                                     win32pipe.PIPE_TYPE_MESSAGE |
                                     win32pipe.PIPE_READMODE_MESSAGE |
                                     win32pipe.PIPE_WAIT, 1, 1024 * 8, 1024 * 8, 0, None)


async def echo(websocket):
    pipe = create_pipe()
    win32pipe.ConnectNamedPipe(pipe, None)
    while True:
        data = win32file.ReadFile(pipe, 1024 * 2)
        await websocket.send(data[1])


async def main():
    async with websockets.serve(echo, "0.0.0.0", 7777):
        await asyncio.Future()  # run forever


if __name__ == '__main__':
    asyncio.run(main())


  


  The way I start ffmpeg

  


  .\ffmpeg.exe -f dshow -i audio="Stereo Mix (Realtek High Definition Audio)" -acodec libmp3lame  -ab 320k -f mp3 -probesize 32 -muxdelay 0.01 -y \\.\pipe\__audio_ffmpeg


  


  On the JS side the code is a little bit long, but essentially I am just reading a web socket and appending to buffer

  


  this.buffer = this.mediaSource.addSourceBuffer('audio/mpeg')


  


  Also as you see I tried to use -probesize 32 -muxdelay 0.01 flags, but no luck as well

  


  I tried to use plain tag as well, but still - this couple-of-seconds delay exists

  


  What can I do ? Am I missing something ? Maybe I have to disable buffering somewhere ?