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• FFMPEG Video Multiplexer [closed]

  24 janvier, par Nader

  I am a DirectShow developer, I used to build multiplexers that take 2 video inputs and generate one output, I would then use a video encoder mux to feed it the output + anothrr audio stream to generate the final video output. The multiplexer (DirectShow framework) allows me to process the input video from two sources (for example, adding effects using the two frames).

  


  How can this can be done using FFMPEG ?

  


• ffmpeg won't execute properly in google app engine standard nodejs

  3 septembre 2019, par tommyc38

  I have tried for three full days to get GAE (standard - nodejs) to run a simple video transcoder from MOV to MP4 using ffmpeg. I have tried using ffluent-ffmpeg, kicking off a child process (e.g. spawn), and nothing works. As soon as it hits the call to the executable it always errors. I have confirmed ffmpeg is installed and even tried using ffmpeg-static. Moreover, I have it working on my local machine with no problems (using all of the aforementioned ways).

  I have also tried logging the errors and nothing is really all that helpful. I can see its working through any installed package including ffmpeg (system package).

  Below is the pseudo code...step three is where the problem occurs.

  1. Send file name to GAE endpoint
  2. Download the file from google cloud storage to a temp file
  3. Transcode using ffmpeg
  4. Upload temp file to google cloud storage
  5. Remove old google cloud storage file
  6. Remove temp file

  The file I am using to test is 6MB...a 5 second video I took on my iPhone. Thank you in advance.

  UPDATE : I successfully deployed the exact same code to Node Flex environment and everything works great. I wasn’t able to get any errors in the standard environment that directed me where to look but my guess is it has something to do with how it stores the file I pipe into FFMPEG on GAE Node Standard. The docs say its a virtual file system that uses RAM. I’d love to hear if anybody managed to get it working in the standard environment.

• Create and update HLS playlist programmatically

  26 février 2018, par Pierre P.

  I have a C++ application that records audio from my default input device, encodes it in AAC format and writes to a .aac file. I want to use HTTP Live Streaming to live stream this AAC file. According to this question, I have to create a FFMPEG script to split my audio file into several .ts files.

  # bitrate, width, and height, you may want to change this
  
  BR=512k
  
  WIDTH=432
  
  HEIGHT=240
  
  input=${1}
  
  
  
  # strip off the file extension
  
  output=$(echo ${input} | sed 's/\..*//' )
  
  
  
  # works for most videos
  
  ffmpeg -y -i ${input} -f mpegts -acodec libmp3lame -ar 48000 -ab 64k -s ${WIDTH}x${HEIGHT} -vcodec libx264 -b ${BR} -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 7 -trellis 0 -refs 0 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate ${BR} -bufsize ${BR} -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 30 -qmax 51 -qdiff 4 -level 30 -aspect ${WIDTH}:${HEIGHT} -g 30 -async 2 ${output}-iphone.ts

  (It is slightly different in my case because I only work with audio)

  Can I do so programmatically in C++ and if so, do I have to use a third-part library or does macOS provide native functions to do so ?