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• ffmpeg concat image with video, but output extra long

  11 octobre 2020, par Morris

  My goal : To concat an image (foo.jpg) with a video (bar.mp4, 3 seconds long). Show foo.jpg for 2 seconds only. Output video should be just around 5 seconds long.

  


  I used :

  


  ffmpeg -loop 1 -t 2 -framerate 1 -i foo.jpg -f lavfi -t 2 -i anullsrc -i bar.mp4 -filter_complex "[0][1][2:v][2:a] concat=n=2:v=1:a=1 [vpre][a];[vpre]fps=24,scale=32:24[v]" -map "[v]" -map "[a]" out.mp4


  


  I think the command means :
Loop foo.jpg for 2 seconds at framerate of 1 frame per second. At the same time, add silent audio track to the 2-second foo.jpg video.
Then concat with bar.mp4.
Make final output framerate 24 fps. Scale it to 32x24 dimension (intentionally tiny for testing).

  


  Expected : output to be about 5 seconds long in total.

  


  Reality : output is 3 minutes and 38 seconds long. The first 5 seconds is perfect. After that, video just stays silent with the 5th-second frame frozen until the end.

  


  My research shows that it might be related to -video_track_timescale

  


  This command also fails with same over-long result (I added -video_track_timescale 600) :

  


  ffmpeg -loop 1 -t 2 -framerate 1 -i foo.jpg  -f lavfi -t 2 -i anullsrc -i bar.mp4 -filter_complex "[0][1][2:v][2:a] concat=n=2:v=1:a=1 [vpre][a];[vpre]fps=24,scale=32:24[v]" -map "[v]" -map "[a]" -video_track_timescale 600 out.mp4


  


  Additional info about file bar.mp4 :

  


  $ ffmpeg -i bar.mp4 
ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 11.0.3 (clang-1103.0.32.62)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_1 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  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
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'bar.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.29.100
  Duration: 00:00:01.94, start: 0.000000, bitrate: 641 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, smpte170m/unknown/smpte170m), 480x360 [SAR 1:1 DAR 4:3], 354 kb/s, 24.58 fps, 24.58 tbr, 113734695.00 tbn, 49.16 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 280 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
At least one output file must be specified


  


• How to process remote audio/video stream on WebRTC server in real-time ? [closed]

  7 septembre 2020, par Kartik Rokde

  I'm new to audio/video streaming. I'm using AntMedia Pro for audio/video conferencing. There will be 5-8 hosts who will be speaking and the expected audience size would be 15-20k (need to mention this as it won't be a P2P conferencing, but an MCU architecture).

  


  I want to give a feature where a user can request for "convert voice to female / robot / whatever", which would let the user hear the manipulated voice in the conference.

  


  From what I know is that I want to do a real-time processing on the server to be able to do this. I want to intercept the stream on the server, and do some processing (change the voice) on each of the tracks, and stream it back to the requestor.

  


  The first challenge I'm facing is how to get the stream and/or the individual tracks on the server ?

  


  I did some research on how to process remote WebRTC streams, real-time on the server. I came across some keywords like RTMP ingestion, ffmpeg.

  


  Here are a few questions I went through, but didn't find answers that I'm looking for :

  


  

  1. Receive webRTC video stream using python opencv in real-time

  


  2. Extract frames as images from an RTMP stream in real-time

  


  3. android stream real time video to streaming server

  


  


  I need help in receiving real-time stream on the server (any technology - preferable Python, Golang) and streaming it back.

  


• mencoder. Encoding from multiple input image files compatible with web browser (No video support and MIME type) [duplicate]

  23 juin 2020, par iblasi

  I have multiple JPG files that I want to use to make a TimeLapse video compatible with the web browser to upload it on my web page.
Create a video with mencoder from multiple images is explained in some webpages such us here, that shows how to create a video.

  


  ls -Ltr my_Pics/*.jpg >files.txt
mencoder -nosound -ovc lavc -lavcopts vcodec=mpeg4 -o video.avi -mf type=jpeg:fps=4 mf://@files.txt


  


  The video is set with no sound and to have one picture every 250ms (4 fps).
These command lines create an AVI video that I can see correctly with the VLC video tool. However, if I try to open it in a web browser it shows an error :

  


  


  No Video with Supported Format and MIME type found

  


  


  So, based on other similar comments (as here), I tryed to use ffmpeg renaming all my files as ffmpeg requires a number serial format. But it happens the same, that I can see it in VLC but not in the browser.

  


  ffmpeg -r 4 -i ./output/%04d.jpg -vcodec libx264 video.mp4


  


  Based on research made on internet I am quite sure that it is due the the encoding and/or container. I tryed multiple options of codecs nd containers existing on documentation (here) but still not able to find a way to work.

  


  If, once I create the video, I use the VLC tool to manually convert the video to ".m4v" I was able to create a video that the web browser recognizes. But I would like to do it with command lines to automate it.