Recherche avancée

Sur d’autres sites (9544)

• Update Database After shell_exec() Using FFMPEG/Laravel

  25 avril 2020, par Mike

  I want to update a database record (MySQL) after the FFMPEG process is complete. Normally I would just call a function just after the FFMPEG command, but I needed to add > /dev/null 2>/dev/null & so I didn't have to wait for the FFMPEG process to complete which would mean the REST call would hang for a LONG time and that would cause bad UX for the front-end.

  



  I'm not sure where to begin with it. My first thought is to make a REST request, but maybe calling the method in Laravel directly would be better.

  



  Can I do a curl call after the FFMPEG command ? Or can I call a PHP method ? Or is there a better way ?

  



  PHP FFMPEG Method

  



  private function transcode($movie)
{
    try {
        $name       = 'master.' . $movie->extension;
        $this->path = storage_path('app/public/movies/') . $movie->id . '/';
        $fps        = $this->getFrameRate($name);
        $width      = $this->getVideoWidth($name);
        $height     = $this->getVideoHeight($name);

        // ffmpeg commands
        $c = $this->buildCommand($width, $height, $fps);

        // ffmpeg -  added '> /dev/null 2>/dev/null &' so it will not wait to finish
        $ffmpeg = shell_exec('ffmpeg -i ' . $this->path . $name . ' -progress ' . $this->path . 'transcode.log' . $c . ' > /dev/null 2>/dev/null &');

        return response()->json(['message' => 'transcode initiated'], 200); 

    }
    catch(\Exception $e)
    {
        return response()->json(['error' => $e->getMessage()], 500);
    }
}


  



  Here is the bash curl post idea

  



  $curl = 'curl --data "param1=value1&param2=value2" http://hostname/transcode/complete';

$ffmpeg = shell_exec('ffmpeg -i ' . $this->path . $name . ' -progress ' . $this->path . 'transcode.log' . $c . '; $curl > /dev/null 2>/dev/null &');


  



  I could be going down the wrong path with the above, but I'm trying to move forward with something.

  


• 3 Hours video opens as 3 minutes [ffmpeg][vegas][Windows]

  20 avril 2018, par CrazyFrog

  I have a mp4 video that is 3:55:00 long. However, when I try to edit it with Vegas 13 or ffmpeg, the softwares only find the first 00:03:16 part.

  I tried to search for similar questions, but did not find any answers.

  Video details(Windows explorer) :
  Length : 03:55:00
  Res : 1280x720
  Data rate:1342 kbps
  Total bitrate : 1408 kbps
  Frame rate : 25 f/s

  Audio
  Bit rate:65 kbps
  Channels : 2(stereo)
  Audio sample rate : 44.1 kHz

  Video details(Vegas, ffmpeg) :
  Format : AVC
  ffmpeg compatible_brands : isomiso2avc1mp41

  I tried using ffmpeg to convert it to a regular mp4. Unfortunately it only finds the first 3:16 minutes too.

  UPDATE

  .\ffmpeg.exe -i origin.mp4 -c copy test.mp4
  
  
  
  ffmpeg version N-88514-gd5995c531d Copyright (c) 2000-2017 the FFmpeg developers
  
  built with gcc 7.2.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth --enable-libmfx
  
    libavutil      56.  0.100 / 56.  0.100
  
    libavcodec     58.  1.100 / 58.  1.100
  
    libavformat    58.  0.102 / 58.  0.102
  
    libavdevice    58.  0.100 / 58.  0.100
  
    libavfilter     7.  0.101 /  7.  0.101
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'origin.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
    Duration: 03:55:00.64, start: 0.000000, bitrate: 1414 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, 1343 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 65 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler
  
  Output #0, mp4, to 'test.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf58.0.102
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, q=2-31, 1343 kb/s, 25 fps, 25 tbr, 12800 tbn, 12800 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 65 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #0:1 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  frame= 4921 fps=0.0 q=-1.0 Lsize=   26141kB time=00:03:16.72 bitrate=1088.6kbits/s speed=1.68e+003x
  
  video:24412kB audio:1569kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.619153%

• 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 ?