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• Convert fla and swf to mp4 video

  4 mai 2021, par Max Williams

  I have an .fla file and a .swf file which were intended to make a Flash Video some years ago.

  


  I want to make an mp4. i tried simply passing the swf into ffmpeg, and the resulting mp4 file had the audio but no video. If I use the fla and the swf, would I get video too ? I don't know how Flash worked wrt the different file types.

  


  EDIT : as asked for, the output from doing just the swf and from using both as the input :

  


  swf alone :

  


  $ ffmpeg -i input.swf  output.mp4
ffmpeg version git-2018-05-03-0f0d468 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.4)
  configuration: --enable-gpl --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-version3
  libavutil      56. 18.100 / 56. 18.100
  libavcodec     58. 19.100 / 58. 19.100
  libavformat    58. 13.100 / 58. 13.100
  libavdevice    58.  4.100 / 58.  4.100
  libavfilter     7. 21.100 /  7. 21.100
  libswscale      5.  2.100 /  5.  2.100
  libswresample   3.  2.100 /  3.  2.100
  libpostproc    55.  2.100 / 55.  2.100
[swf @ 0x22483c0] SWF compressed file detected
[swf @ 0x22483c0] Estimating duration from bitrate, this may be inaccurate
Input #0, swf, from 'input.swf':
  Duration: 00:00:51.87, bitrate: 112 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, mono, fltp, 112 kb/s
File 'output.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
Output #0, mp4, to 'output.mp4':
  Metadata:
    encoder         : Lavf58.13.100
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 69 kb/s
    Metadata:
      encoder         : Lavc58.19.100 aac
[swf @ 0x22483c0] New audio stream 0:1 at pos:226047 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:2 at pos:234457 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:3 at pos:245057 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:4 at pos:255657 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:5 at pos:270637 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:6 at pos:281967 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:7 at pos:290742 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:8 at pos:300612 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:9 at pos:309022 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:10 at pos:463819 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:11 at pos:696822 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:12 at pos:709202 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:13 at pos:735377 and DTS:0s
[swf @ 0x22483c0] New audio stream 0:14 at pos:745561 and DTS:0s
size=     265kB time=00:00:31.04 bitrate=  69.8kbits/s speed= 109x    
video:0kB audio:259kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.325915%
[aac @ 0x22a06c0] Qavg: 1972.990



  


  both :

  


  $ ffmpeg -i input.fla -i input.swf output.mp4
ffmpeg version git-2018-05-03-0f0d468 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.4)
  configuration: --enable-gpl --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-version3
  libavutil      56. 18.100 / 56. 18.100
  libavcodec     58. 19.100 / 58. 19.100
  libavformat    58. 13.100 / 58. 13.100
  libavdevice    58.  4.100 / 58.  4.100
  libavfilter     7. 21.100 /  7. 21.100
  libswscale      5.  2.100 /  5.  2.100
  libswresample   3.  2.100 /  3.  2.100
  libpostproc    55.  2.100 / 55.  2.100
input.fla: Invalid data found when processing input


  


• HTML5 Video Compatibility (MP4, WEBM, OGG) in 2021

  19 juillet 2021, par William

  The support of HTML5 video has evolved a lot over the years. I am trying to understand whether the <video></video> element still needs to have three sources : MP4, WEBM, and OGG.

  


  There are a lot of answers throughout StackOverflow with deeply conflicting information - some of which say that you just need MP4 now, others say, MP4 and WEBM are enough, and then finally many say that you need all three (although many of those article are 10 years old).

  


  W3 suggests that either MP4 or WEBM alone would have universal support (Even though I found a 2011 article from Google saying that they would be removing support for MP4/H.264). Wikipedia paints a more complicated picture (as well as listing that Google Chrome does indeed support MP4/H.264). Azure Media services ONLY seems to allow output in MP4, which would suggest to me that MP4 must have widespread compatibility.

  


  Also see Example 1, Example 2, Example 3.

  


  Is there any definitive information on what video types to include in an HTML5 video player to achieve widespread compatibility ?

  


  Background : I am building a Content Management Platform that allows uploading videos. When a new video is uploaded, a conversion process kicks off to convert the video into the required formats. This takes time and CPU/Memory, so if it is possible I would like to convert uploaded videos into as few formats as possible.

  


  p.s. This question HAS been asked before, however, the fundamentals of playing video on the web continually evolve and most of the answers out there have become irrelevant.

  


• ffmpeg libavformat - output method

  2 octobre 2022, par Branqueira

  I have made a website that can display x camera live using rtsp.

  


  The user connects to the url, the javascript hls.js makes a request to the tomcat java server. If this is the first invocation, the code does jni to call my c code which uses ffmpeg.
In the ffmpeg part, the input is the rtsp : // url and the output is the hls + http stream which corresponds to my tomcat server. This allows to put in a buffer if there is x connection to see the ip camera.
In conclusion, there is only one ffmpeg instance running regardless of the number of people calling the html page.

  


  All this works well and for 2 years already.

  


  But I would like to improve the "machin"

  


  I would like the output of ffmpeg to always be in hls but instead of it being http, I want this to invoke a method (for example) that contains the file's buffer and its name.

  


  for that I use

  


  avformat_alloc_output_context2 (& formatOutputCtx, NULL, "hls", output_url);


  


  which allows me to select the hls + http protocol automatically

  


  I looked at the doc but couldn't find a solution using another protocol or doing one custom.

  


  If someone has an idea ? otherwise no big deal :)