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• FFMPEG tpad and AF not working with incorrect duration

  9 octobre 2020, par Devin Dixon

  I am live streaming via rtmp with ffmpeg, and I have tpad and AF to add padding to the video and audio of a file.

  


  My version of ffmpeg

  


  ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --prefix=/root/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/root/bin --enable-libfdk-aac --enable-gpl --enable-libass --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libtheora --enable-openssl --enable-nonfree
libavutil      56. 31.100 / 56. 31.100
libavcodec     58. 54.100 / 58. 54.100
libavformat    58. 29.100 / 58. 29.100
libavdevice    58.  8.100 / 58.  8.100
libavfilter     7. 57.100 /  7. 57.100
libswscale      5.  5.100 /  5.  5.100
libswresample   3.  5.100 /  3.  5.100
libpostproc    55.  5.100 / 55.  5.100


  


  An example command :

  


  ffmpeg  -re -i https://my-cdn.com/video.mp4 -filter_complex "tpad=start_duration=10:stop_duration=15" -af "adelay=10000|10000" -maxrate 2M -crf 24 -bufsize 6000k -c:v libx264 -preset superfast -tune zerolatency -strict -2 -c:a aac -ar 44100 -f flv rtmp://stream.example.com/live/some-channel-id


  


  Now it works for as intended for many of the video, except sometimes I will get :

  


  [flv @ 0x559c59067700] Failed to update header with correct duration.170.4kbits/s speed=1.04x    
[flv @ 0x559c59067700] Failed to update header with correct filesize.


  


  When this happens the "tpad=start_duration=10:stop_duration=15" -af "adelay=10000|10000" are ignored and no longer take affect.

  


  Has anyone else had this issue ? And do you now how to resolve it ?

  


  Or during a conversion process(we've been converting mov to mp4), is there a way to ensure the file size and duration is correct ?

  


• FFMPEG and libx264, WMP and QT playback at different res/aspect than specified

  10 septembre 2013, par Foltz

  So, for a little context. I'm trying to convert a JPEG2000 in .mxf video to an h264 in .mp4, maintaining its resolution/frame-size/aspect ratio. I'll say up front I'm not 100% sure on how all these ratios tie together.

  So, my input file is 720x486. I have a sample h264 file with specs I'm trying to emulate (from an OpenCube video ingest machine) with ffmpeg. The OpenCube file plays back at 720x486 in both Quick Time 7 and Windows Media Player 12. It reports a PAR of 9:10 and DAR of 4:3 (Using VideoSpec to see these values).

  When I make my derivative h264 using ffmpeg I can run the command :

  ffmpeg -i lac-test.mxf -vcodec libx264 -s 720x486 -aspect 4:3 -b:v 2500000 -pix_fmt yuv420p -coder 0 -acodec libfaac -b:a 256000 lac-test.mp4

  The output file reports the correct values in VideoSpec (720x486, DAR 4:3, PAR 9:10), but plays back in QT at 648x486. Weirder, the video plays back at the proper resolution (that is, same size as the sample OpenCube file).

  Now, an alternative command creates yet another predicament :

  ffmpeg -i lac-test.mxf -vcodec libx264 -s 720x486 -b:v 2500000 -pix_fmt yuv420p -coder 0 -acodec libfaac -b:a 256000 lac-test.mp4

  So omitting the aspect ratio gives me an output file that plays at 720x486 in QT7 (good !) but reports a DAR of 1.481 and an Undefine PAR. Unfortunately, THIS file plays back slightly too wide in WMP12.

  I've tried a slew of different strings to try and get something that plays back correctly. I've tried using various filters with the -fv flag such as :

  ffmpeg -i lac-test.mxf -vcodec libx264 -vf "scale=720x486,setsar=9:10,setdar=4:3" -b:v 2500000 -pix_fmt yuv420p -coder 0 -acodec libfaac -b:a 256000 lac-test.mp4

  This results in a video that shows all the correct specs in VideoSpec, and plays back correctly in WMP12 (same size as OpenCube file), but still plays back at 648x486 in QT7.

  What gives ?

• ffmpeg : transcode over socket

  19 septembre 2020, par user14258924

  I am consuming multiple streams of h264 video and aac audio data in my own RTMP server and I want to transcode these streams into different codecs so I want to send these tracks, in chunks, into ffmpeg and get the transcoded result back. But due to performance requirements I want to use some kind of streaming where I can just open up a socket to get files in/out instead of dumping the data into individual files and then reading transcoded files back.

  


  Is it possible with ffmpeg ?