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• How to get memory location in C# while using structure of C [on hold]

  12 août 2017, par Nirmal

  I am using C# to record a video from IP camera and I am using libavcodec57 library for that. So if I need to write or read any property of a structure I need to use Marshal class in C#, like

  int width = Marshal.ReadInt32(in_codecctx, 40);

  In above in_codecctx is of type IntPtr which points AVCodecContext structure in C. So there are many properties in that like width, height etc and to read I need to use above line of code and for that I have to know the address of that property in the memory, which is base address of structure plus offset to that property. Base address I know but how to find actual offset of any property.

  One way is, I need to create same structure in C# which is defined in libavcodec library and use below code, this will give me offset of width from structure defined in C#.

  Marshal.OffsetOf(typeof(AVCodec.AVCodecContext),"width")

  But this needs great efforts and still not accurate as it gave me some wrong values.

  ---

  [Update from comment]

  This link is for structure I need to use, is there any way to identify the offset of width and height, if yes how ?

• PHP $_FILE Temp Location open_basedir Conflict PHP-FPM/Nginx

  16 septembre 2017, par xendi

  I have a script for uploading files to my site. It works great but now I’m wanting to extract meta data from video files using the ffmpeg/ffprobe library. My code :

  if (!empty($_FILES)) {
  
              $requestAr = requestCheck(['title', 'description']);
  
              $assetName = randomString(11);
  
              $assetSalt = randomString(3);
  
  
  
              $bucket = "videos";
  
              $fileTmpPath = $_FILES['qqfile']['tmp_name'];
  
              $filenameOrig = $_FILES['qqfile']['name'];
  
              $fileExtension = pathinfo($filenameOrig, PATHINFO_EXTENSION);
  
              $assetFilename = $assetName . "_$assetSalt.$fileExtension";
  
  
  
              $trustedExtensions = ['webm', 'mp4', 'ogg'];
  
              if(array_search($fileExtension, $trustedExtensions) !== false) {
  
                  $ffprobe = FFMpeg\FFProbe::create([
  
                      'ffprobe.binaries' => '/usr/bin/ffprobe'
  
                  ]);
  
                  $fileInfo = $ffprobe
  
                      ->format($fileTmpPath) // extracts file informations
  
                      ->get('duration');             // returns the duration property
  
                  print_r($fileInfo);
  
              }
  
  }

  I wind up with this error :

  file_exists(): open_basedir restriction in effect. File(/usr/bin/ffprobe) is not within the allowed path(s): <lists all="all" the="the" directories="directories" in="in" variable="variable">
  
  </lists>

  I have passed the ffmpeg library the absolute path to ffprobe so it knows where it is. I was searching around and some people were saying this is because the lib can’t access the tmp directory with the uploaded file. In either case, I’ve been trying to disable open_basedir or at least add the paths I need to it to get this to work.

  I tried setting open_basedir to none in my php.ini but it didn’t work. When I view phpinfo(), it still lists a bunch of paths for that setting. I tried to grep where open_basedir exists on the server. Indeed, it’s the php.ini files and I shouldn’t need to modify any other than the one reported in phpinfo().

• FFMPEG : merge multiple audio files to one but audio parts should be on specific location

  20 octobre 2017, par Nihad

  I am trying to build one long audio file from multiple small audio files but these parts should start at a specific time.

  This is a command that I am using : (base audio is 135s long)
  ffmpeg -y -i base.mp3
  
   i first.wav
  
   i second.wav
  
   filter_complex
  "[1] adelay=90068|90068[a1] ;
  [2] adelay=130168|130168[a2] ;
  [0] [a1] [a2] amix=3" output.mp3

  The command is working as it should, output file is being created but the delay is not as specified. The first small file has delay of 87000 (3s earlier) and the second audio has delay of 126000 (4s earlier).

  This is the ffmpeg output :
  ffmpeg version 3.0.1 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (GCC)
  configuration : —enable-gpl —enable-version3 —disable-w32threads —enable-avisynth —enable-bzlib —enable-fontconfig —enable-frei0r —enable-gnutls —enable-iconv —enable-libass —enable-libbluray —enable-libbs2b —enable-libcaca —enable-libdcadec —enable-libfreetype —enable-libgme —enable-libgsm —enable-libilbc —enable-libmodplug —enable-libmp3lame —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libopenjpeg —enable-libopus —enable-librtmp —enable-libschroedinger —enable-libsoxr —enable-libspeex —enable-libtheora —enable-libtwolame —enable-libvidstab —enable-libvo-amrwbenc —enable-libvorbis —enable-libvpx —enable-libwavpack —enable-libwebp —enable-libx264 —enable-libx265 —enable-libxavs —enable-libxvid —enable-libzimg —enable-lzma —enable-decklink —enable-zlib
  libavutil 55. 17.103 / 55. 17.103
  libavcodec 57. 24.102 / 57. 24.102
  libavformat 57. 25.100 / 57. 25.100
  libavdevice 57. 0.101 / 57. 0.101
  libavfilter 6. 31.100 / 6. 31.100
  libswscale 4. 0.100 / 4. 0.100
  libswresample 2. 0.101 / 2. 0.101
  libpostproc 54. 0.100 / 54. 0.100
  [mp3 @ 059b4400] Skipping 0 bytes of junk at 227.
  Input #0, mp3, from ’base.mp3’ :
  Metadata :
  encoder : Lavf57.25.100
  Duration : 00:02:21.04, start : 0.025057, bitrate : 31 kb/s
  Stream #0:0 : Audio : mp3, 44100 Hz, mono, s16p, 32 kb/s
  Guessed Channel Layout for Input Stream #1.0 : mono
  Input #1, wav, from ’first.wav’ :
  Duration : 00:00:14.88, bitrate : 352 kb/s
  Stream #1:0 : Audio : pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 1 channels, s16, 352 kb/s
  Guessed Channel Layout for Input Stream #2.0 : mono
  Input #2, wav, from ’second.wav’ :
  Duration : 00:00:02.57, bitrate : 352 kb/s
  Stream #2:0 : Audio : pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 1 channels, s16, 352 kb/s
  Output #0, mp3, to ’output.mp3’ :
  Metadata :
  TSSE : Lavf57.25.100
  Stream #0:0 : Audio : mp3 (libmp3lame), 22050 Hz, mono, fltp (default)
  Metadata :
  encoder : Lavc57.24.102 libmp3lame
  Stream mapping :
  Stream #0:0 (mp3) -> amix:input0
  Stream #1:0 (pcm_s16le) -> adelay
  Stream #2:0 (pcm_s16le) -> adelay
  amix -> Stream #0:0 (libmp3lame)
  Press [q] to stop, [?] for help
  size= 384kB time=00:01:38.11 bitrate= 32.0kbits/s speed= 196x
  
  [libmp3lame @ 059eef60] Trying to remove 576 samples, but the queue is empty
  size= 551kB time=00:02:21.03 bitrate= 32.0kbits/s speed= 191x
  
  video:0kB audio:551kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead : 0.040223%

  Is there a way to fix this problem, maybe with some additional parameter or something ?

  Thanks :)