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• How to use the result of guardian-ffmpeg-Android ?

  4 avril 2013, par user1914692

  I am trying to build ffmpeg on Android. There are many tutorials. Some are very old.
  So I want to try one that can use newer version of ffmpeg and Android NDK.
  After long time searching, I find one, guardianproject / android-ffmpeg

  The project was updated several months ago.
  NDK r8 is used. ffmpeg is put from online, so a latest version.

  After I follow all the instruction, I am confused which result I should use, and how to use it.
  The README mentions testing, like :

  # embedding metadata into a matroska video /data/local/ffmpeg -y -i test.mp4 \
  
      -attach attach.txt -metadata:s:2 mimetype=text/plain \
  
      -acodec copy -vcodec copy testattach.mkv

  First, I fail to find the path : /data/local
  Second, this is a command. How will I use it in Android ?

  Totall confused.
  Any light ?

• Record linux screen (audio/video) and stream to an API

  5 juin 2024, par Joel

  I'm running a browser on a Linux server, and I'm trying to figure out the best way to capture the audio/video of the screen, and stream that with as little latency as possible to an API on another server.

  


  Requirements :

  


  

  • Stream the audio from server A to server B so that server B can pipe that forward to an online transcription service.

  


  • Stream the audio and video from server A to server B so that server B can store the contents to some kind of blob storage. If the stream is killed for some reason before it's ended, the partial contents should be saved and still be playable.

  


  


  I've successfully captured the audio/video to disk using ffmpeg (see below), but I'm really unsure how I should change that to stream the contents to an API instead.

  


  The API is currently ASP.NET Core 8, but it could be node/python/whatever really.

  


  ffmpeg -y -video_size 1920x1080 -framerate 30 -f x11grab -i :0 -f pulse -i default -c:v libx264 -pix_fmt yuv420p -c:a aac -strict experimental /app/recordings/output.mp4


  


• Merge audio (m4s) segments into one

  20 avril 2022, par akinuri

  I recently started learning Laravel, and currently watching an online course. Online courses are fine, but I like to have local copies, so I'm trying to download/merge segmented audio from Laracasts : Laravel 8 From Scratch series.

  


  I've written some scripts (in Python) that does the following :

  


  

  1. Download the master.json

  


  2. Read master.json and download audio segments

  


  3. Merge the segments into a single file (the file is not playable yet)

  


  4. Process the audio file via ffmpeg (now it's playable, but has issues)

  


  


  I think there's a problem with the step 3 and/or 4.

  


  In step/script 3, I create a new file, and add the contents of the segments to the file in binary.

  


  Then (step/script 4), run a ffmpeg command in python : ffmpeg -i merged-file.mp4 -c copy processed-file.mp4

  


  However, the final file doesn't work/play as expected. There's a delay in the beginning, and some parts seem to be cut off/skipped.

  


  There are three possibilities :

  


  

  • Segment files are problematic (not likely ?)

  


  • I'm doing the merging wrong

  


  • I'm doing the ffmpeg processing wrong

  


  


  Can someone guide me here ?

  


  ---

  


  The issues/colored parts in the ffmpeg output are :

  


  ...
[mov,mp4,m4a,3gp,3g2,mj2 @ 000001cfbc0de780] could not find corresponding track id 2
[mov,mp4,m4a,3gp,3g2,mj2 @ 000001cfbc0de780] could not find corresponding trex (id 2)
...
[aac @ 000001cfbc0f0380] Number of bands (31) exceeds limit (6).
...
[mp4 @ 000001cfbc20ecc0] track 0: codec frame size is not set
...
[mp4 @ 000001cfbc20ecc0] Non-monotonous DTS in output stream 0:0; previous: 318318, current: 286286; changing to 318319. This may result in incorrect timestamps in the output file.
...


  


  Everything required for a test case is located in GitHub (akinuri/dump/m4s-segments/). Screenshot of the contents :

  


  

  


  ---

  


  Note : there are two types/formats of audio in the master.json : mp42 and dash. dash works as expected, and seem to be used in limited videos/courses. On the other hand, mp42 appears more. So I need a way to make mp42 work.