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• After using ffmpeg to remove some streams in mkv file, it takes much longer to open the media file in potplayer in Windows

  31 mars 2024, par r ne

  It is well known that ffmpeg command line can be used to remove some streams in mkv or mp4 files, and I have no problem in doing so.

  


  ffmpeg -i input.mp4 -map 0 -map -0:a:3 -map -0:a:6 -c copy output.mp4


  


  The only issue I meet sometimes is, the result media files in mkv or mp4 are much much slower to be opened by player. I cannot figure out the real reason and bypass it.

  


  Any hint or help on solving this issue ? Thanks !

  


• file concat mp4's using ffmpeg does not concat

  3 juillet 2019, par bozzmob

  I have 3 mp4’s in my input text file(a.txt).

  Contents of a.txt-

  file '/Users/path-to-file/1.mp4'
  
  file '/Users/path-to-file/2.mp4'
  
  file '/Users/path-to-file/3.mp4'

  The command I am executing (Referring to Docs and StackOverflow) -

  ffmpeg -safe 0  -f concat -i a.txt -c copy output.mp4

  The output.mp4 is created. When I play the video output.mp4, I see the video & audio of 1.mp4 playing. But, after it completes, I just hear the audio of 2.mp4 and 3.mp4. The video is stuck on the last frame of 1.mp4 during that time.

  Please can you tell me as to what I am missing ?

  ---

  Update :
  The video plays only in quick-view of quicktime player. The video crashes VLC when I try to open in VLC. I think this suggests the encoding is not right or something else is wrong in the above steps.

• FFMPEG decode from RTP dump file into mp3 file

  5 février 2021, par pingvincible

  I'm trying to save RTP stream into mp3 file. I use this command :

  


  ffmpeg -loglevel debug -protocol_whitelist file -f rtp -i microphone.rtpdump -f mp3 microphone.mp3


  


  I get this result :

  


  user@pc:~/$ ffmpeg-amrnb -loglevel debug -protocol_whitelist file -f rtp -i microphone.rtpdump -f mp3 microphone.mp3
ffmpeg version N-100958-g4f3d8cb554 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
  configuration: --enable-gpl --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-nonfree --enable-version3
  libavutil      56. 64.100 / 56. 64.100
  libavcodec     58.120.100 / 58.120.100
  libavformat    58. 65.101 / 58. 65.101
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7.102.100 /  7.102.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'rtp'.
Reading option '-i' ... matched as input url with argument 'microphone.rtpdump'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
Reading option 'microphone.mp3' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url microphone.rtpdump.
Applying option f (force format) with argument rtp.
Successfully parsed a group of options.
Opening an input file: microphone.rtpdump.
[rtp @ 0x556947200580] Unable to receive RTP payload type 97 without an SDP file describing it
microphone.rtpdump: Invalid data found when processing input


  


  It looks like that microphone.rtpdump file format is correct as ffmpeg can find RTP payload type 97. The problem is that I don't understand how to use SDP file in this situation.

  


  I have an SDP file for this payload type which I use, when I send data over network. It looks like this :

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.65.101
m=audio 1234 RTP/AVP 97
b=AS:12
a=rtpmap:97 AMR/8000/1
a=fmtp:97 octet-align=1


  


  And now I want to decode RTP stream from file, not by receiving it from network.

  


  How to adapt my SDP file to read RTP stream from file ?

  


  UPDATE : My rtpdump file is not a real rtpdump file format. It is just payloads from UDP packets written into file without any additional headers.