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• python/video - removing frames without re-encoding

  18 février 2017, par nxet

  I understand this might be a trivial question, but so far I had no luck with the various solutions I tried and I'm sure there must be a convenient way to achieve this.

  How would I proceed removing frames/milliseconds from a video file without slicing, merging and re-encoding?
  All solutions I found involved exporting various times to various formats, and I'm hopeful there will be no need to do so.

  With ffmpeg/avconv it's necessary to convert the temporary streams to .ts, then concatenate, and finally re-encode in the original format.

  Python library MoviePy seemed to do quite exactly what I needed but:

  • The cutout function returns a file which can not be exported, as the write_videofile function tries and fails to fetch the removed frames
  • If I instead slice the original file into various clips and then merge them with concatenate_videoclips, the export doesn't fail but takes twice the length of the video. The resulting video has then a faster frame-rate, with only the cue points for the concatenated videos being timely placed, and audio playing at normal speed. It's also worth noting that the output file, despite being 5-7% shorter in duration, is about 15% bigger.

  Is there any other library I'm not aware of I might look into?
  What I'm trying to imagine is an abstraction layer providing easy access to the most common video formats, giving me the ability to pop the unwanted frames and update the file headers without delving into the specifics of each and every format.

• ffmpeg live stream multiple outputs to same pipe (stdout Python)

  18 février 2017, par WobbaFetttttt

  So I am writing a stream ingest in Python with ffmpeg and want my Python script to receive the video and audio feeds separately without making a temporary file.

  I have tried these subprocess PIPE commands inspired by THIS BLOG and THIS FFMPEG TUTORIAL for Python integration and multiple ffmpeg outputs:

  from subprocess import Popen, PIPE
  
  stream = Popen([
      'ffmpeg',
      '-i', URL, '\\',
  
      '-f', 'image2pipe',
      '-pix_fmt', 'rgb24',
      '-vcodec', 'rawvideo', '-', '\\',
  
      '-f', 's16le',
      '-ac', '2',
      '-ar', '44100',
      '-vcodec', 'pcm_s16le', '-'
  ], stdout=PIPE, stderr=PIPE, bufsize=10**8)
  
  print 'ingesting stream'
  for i in range(300):
      video = stream.stdout.read(640*360*3)
      video = numpy.fromstring(video, dtype='uint8')
      video = video.reshape((360,640,3))
  
      stream.stdout.flush()
  
      audio = stream.stdout.read(2*44100*2/10)
      audio = numpy.fromstring(audio, dtype='int16')
      audio = audio.reshape((len(audio)/2, 2))
  
      stream.stdout.flush()
  
  stream.terminate()
  

  but this gives first pipe read seems to read no data:

  Traceback (most recent call last):
    File "test.py", line 63, in 
      video = video.reshape((360,640,3))
  ValueError: cannot reshape array of size 0 into shape (360,640,3)
  

  I tried running each pipe individually and they work just fine

• Converting Video to mp3 node.js

  18 février 2017, par Adhit Lokesh

  I'm trying below code. can anyone tell me whats wrong in the code

       var ffmpeg = require('fluent-ffmpeg');
  
       var proc = new ffmpeg({ source: 'C:/Users/Public/Videos/Sample 
  
       Videos/Wildlife.wmv', nolog: true })
  
       proc.setFfmpegPath("C:\\nodejs\\ffmpeg")
  
  
       .toFormat('mp3')
  
       .on('end', function() {
       console.log('file has been converted successfully');
       })
       .on('error', function(err) {
       console.log('an error happened: ' + err.message);
       })
       // save to file <-- the new file I want -->
       .saveToFile('C:/Users/Public/Videos/Sample Videos/Wildlife.mp3');
  

  C:\nodejs>node test.js an error happened: spawn C:\nodejs\ffmpeg ENOENT

• Have anyone used ffmpeg with iOS Swift app ? [on hold]

  18 février 2017, par Varun Raj

  I'm trying to build an app that can convert recorded videos from MP4 to webm for better size and quality. I wanted to use FFMPEG and VP9 for it, but not sure if it's technically possible.

  Using Swift 3 and Xcode 8.

  AVCapture for recording the video and I wanted the recorded video to be converted to webm and then upload it to S3 Bucket.

  Now I'm just stuck with the conversion part of it.

• Compile FFmpeg with libfdk_aac

  18 février 2017, par Toydor

  I been reading on how to convert mp3 to m4a, and found that I must compile FFmpeg if I'll use the AAC encoder, libfdk_aac.

  But reading FFmpeg guide on how to compile FFmpeg with libfdk_aac makes no sense for a beginner like me.

  To use libfdk_aac the encoding guide says:

  Requires ffmpeg to be configured with --enable-libfdk_aac --enable-nonfree.

  Where do I put those flags?

  Do I put it here somewhere?:

  cd ~/ffmpeg_sources
  git clone --depth 1 git://github.com/mstorsjo/fdk-aac.git
  cd fdk-aac
  autoreconf -fiv
  ./configure --prefix="$HOME/ffmpeg_build" --disable-shared
  make
  make install
  make distclean
  

  Or maybe here somewhere?

  cd ~/ffmpeg_sources
  git clone --depth 1 git://source.ffmpeg.org/ffmpeg
  cd ffmpeg
  PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig"
  export PKG_CONFIG_PATH
  ./configure --prefix="$HOME/ffmpeg_build" \
    --extra-cflags="-I$HOME/ffmpeg_build/include" --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
    --bindir="$HOME/bin" --extra-libs="-ldl" --enable-gpl --enable-libass --enable-libfdk-aac \
    --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx \
    --enable-libx264 --enable-nonfree --enable-x11grab
  make
  make install
  make distclean
  hash -r
  

  If I'm reading the compile guide right I guess that these two chunks of code is what I need to compile FFmpeg.

  I'm using Ubuntu server 12.4

  UPDATE

  After upgrading my system to Ubuntu 16.04 I had to install ffmpeg again. I still needed libfdk-aac. Fortunately there's a good step-by-step guide at http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu on how to compile ffmpeg.

  I thought I would share how to compile if just interested in compiling ffmpeg with libfdk-aac and libmp3lame.

  If you haven't already a bin in home directory:

  mkdir ~/bin 
  

  Install dependencies. Didn't need the non-server packages:

  sudo apt-get update
  sudo apt-get -y install autoconf automake build-essential libass-dev libfreetype6-dev libtheora-dev libtool libvorbis-dev pkg-config texinfo zlib1g-dev 
  

  Then install the encoders. Had to install yasm as well, otherwise I got errors when compiling.

  sudo apt-get install libfdk-aac-dev
  sudo apt-get install libmp3lame-dev
  sudo apt-get install yasm
  

  Then compile ffmpeg with needed flags

  cd ~/ffmpeg_sources
  wget http://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2
  tar xjvf ffmpeg-snapshot.tar.bz2
  cd ffmpeg
  PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
  --prefix="$HOME/ffmpeg_build" \
  --pkg-config-flags="--static" \
  --extra-cflags="-I$HOME/ffmpeg_build/include" \
  --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  --bindir="$HOME/bin" \
  --enable-libass \
  --enable-libfdk-aac \
  --enable-libfreetype \
  --enable-libtheora \
  --enable-libvorbis \
  --enable-libmp3lame \
  --enable-nonfree \
  --enable-gpl
  PATH="$HOME/bin:$PATH" make
  make install
  make distclean
  hash -r