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• #/bin/sh in one line

  27 juillet 2012, par GoodGuyGreg

  I'm working on some Haskell project using FFmpeg. I need to batch create from a media folder with MP4 files and create screenshots from all of them. I got the code and am using it on a terminal in Unix. It works, but how do I make it in one line to be executed in system "xxxx" in Haskell ?

  If not using several system"xx"...

  #/bin/sh
  
  for i in $(ls *.mp4)
  
  do
  
      ffmpeg -i $i -vframes 7 -y -ss 10 -s 150x150 -an -sameq -f image2 -r 1/5  $i%1d.jpg
  
  done

  I tried :

  import System.Cmd
  
  function = do{system "#/bin/sh";
  
  system "for i in $(ls *.mp4)";
  
  system "do";
  
  system "ffmpeg -i $i -vframes 7 -y -ss 10 -s 150x150 -an -sameq -f image2 -r 1/5  $i%1d.jpg";
  
  system "done";}

  but it gives a error :

  -vframes: No such file or directory
  
  /bin/sh: Syntax error: "done" unexpected

• ffmpeg record audio from Xvfb on Centos

  25 mai 2017, par boygiandi

  I’m trying to record audio from Xvfb. And I have some problems :

  1. What’s the different between alsa and pulse. I get confuse
  2. Server Centos have no soud card :

  arecord -l

  arecord : device_list:268 : no soundcards found...

  3. I may have many Xvfb process, how to record video and audio from specific Xvfb process. I checked this https://trac.ffmpeg.org/wiki/Capture/ALSA#Recordaudiofromanapplication but still don’t understand how it works.

  ffmpeg -f alsa -ac 2 -i hw:0,0 -acodec pcm_s16le output.wav

  I seen many command like this, but I don’t know how to get hw:0,0 ( id of sound card ? )

  Please help. Thanks

• Transcode HLS Segments individually using FFMPEG

  27 mai 2013, par rayh

  I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

  Here is an example ffmpeg command line :

  ffmpeg -threads 1 -nostdin -loglevel verbose \
  
     -nostdin -y -i input.ts -c:a libfdk_aac \
  
     -ac 2 -b:a 64k -y -metadata -vn output.ts

  Inspecting an example sound file shows that there is a gap at the end of the audio :

  

  And the start of the file looks suspiciously attenuated (although this may not be an issue) :

  

  My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

  Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?

  ** UPDATE 1 **

  Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)

  
  

  ** UPDATED 2 **

  So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :

  
  

  I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

  ** UPDATE 3 **

  According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

  For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.