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• target_link_libraries in CMAKE using android studio 2.2.2

  22 novembre 2016, par fadi

  I am facing a weird issue and it’s difficult to know why because the compiler doesnt give any errors.

  I created a new project in android studio 2.2.2 with C++ support.
  I edited the .cpp file inside src/main/cpp and compiled the project to obtain (.so) file that i can use as a shared library. To this point everything works perfect.

  Here is where the problem occurs :

  I am trying to link prebuild shared libraries from ffmpeg. I have already build the libraries in .so format and all I need to do is link them to my .cpp file.

  To link the libraries, I opened the CMakeLists.txt inside android studio and told cmake to link those prebuild shared libraries using the following code :

  add_library(libavformat SHARED IMPORTED)

  set_target_properties(libavformat PROPERTIES IMPORTED_LOCATION  C:/Android  /SDK/MyProjects/ffmpeg_to_jpg/P3/app/src/main/jniLibs/libavformat-55.so)

  include_directories(src/main/cpp/include/)

  target_link_libraries(native-lib  libavformat)

  This code basically links libavformat to native-lib (which is created from my .cpp file)

  The linking process works fine, the reason is because the compiler doesn’t cry about any dependencies.

  However, my original shared library (native-lib) stops working, and by that I mean, I cannot call any functions from within it.

  If i remove the linking line

  target_link_libraries(native-lib  libavformat)

  The native-lib.so works fine and I can call any function from within that does not depend on libavformat.

  I am not sure what is going on, like I said the compiler doesnt issue any warnings or errors. it is almost like after the linking process the content of native-lib is overwritten by libavformat !!!!

  any ideas ?

• ffmpeg's segment_atclocktime cuts at inaccurate times for audio

  3 mai 2023, par Ross Richardson

  I am using ffmpeg's segment format to save files of an AAC stream to disk in hourly segments.
The segmenting works well, but the files are segmented/cut at different times in the clock each hour using segment_atclocktime

  


  I would like each to be exactly on the hour, e.g. 12:00:00, 13:00:00 etc. Or at least, beginning after the hour and not before, e.g. 12:00:00, 13:00:01, 14:00:00 etc.

  


  I am using ffmpeg-python to process the AAC stream and send to two outputs : stdout and these segments.
Here's the code :

  


  out1 = ffmpeg.input(stream, loglevel="panic").output("pipe:",
                                                     format="s16le", 
                                                     acodec="pcm_s16le", 
                                                     ac="1", 
                                                     ar="16000")

out2 = ffmpeg.input(stream, loglevel="info").output("rec/%Y-%m-%d-%H%M%S.aac",
                                                     acodec="copy",
                                                     format="segment",
                                                     segment_time="3600",
                                                     segment_atclocktime="1",
                                                     reset_timestamps="1",
                                                     strftime="1")
            
ffmpeg.merge_outputs(out1, out2)
      .run_async(pipe_stdout=True, overwrite_output=True)


  


  Most files are produced at the desired time : 05:00:00, 06:00:00, 07:00:00, but one or two each day start at 08:59:59 (where 09:00:00 would be desired), or even 16:00:24.

  


  I understand the segment needs to begin on a audio sample so it can't be perfect to the hour, but wondering how I can make it more consistent. Ideally, each hour's recording would begin at 00:00 or later, and not begin before the hour.

  


  I have tried using min_seg_duration 3600, reset_timestamps 1
I am not sure how exactly to use segment_clocktime_wrap_duration for audio, or whether segment_time_delta applies to audio.

  


  I'd appreciate any advice or understanding of how segment_atclocktime works with audio, as much on the internet seems video-focused.

  


• Render SharpDX Texture2D in UWP application

  10 décembre 2019, par Alex

  I’m implementing a solution for hardware-accelerated H264 decoding and rendering in the UWP application. I want to avoid copying from GPU to CPU.
  The solutions consists of 2 parts :

  1. C library that decodes the H264 stream using ffmpeg
  2. UWP/C#/SharpDX application to receive encoded data, pinvoke library and then render decoded frames.

  I receive encoded data in the C# application and send it to the C library to decode and get the pointer to the frame back using pinvoke.

  C part looks good so far. I managed to receive pointer to the decoded frame in GPU in the C library :

  // ffmpeg decoding logic
  
  ID3D11Texture2D* texturePointer = (ID3D11Texture2D*)context->frame->data[0];

  I managed to receive this pointer in C# code and create SharpDX texture from it.

  var texturePointer = decoder.Decode(...data...); // pinvoke
  
  if (texturePointer != IntPtr.Zero)
  
  {
  
      var texture = new Texture2D(texturePointer); // works just perfect
  
  }

  Now I need to render it on the screen. My understanding is that I can create class that extends SurfaceImageSource so I can assign it as a Source of XAML Image object.
  It can be something like this :

  public class RemoteMediaImageSource : SurfaceImageSource
  
  {
  
      public void BeginDraw(IntPtr texturePointer)
  
      {
  
          var texture = new Texture2D(texturePointer);
  
          // What to do to render texture in GPU to the screen?
  
      }
  
  }

  Is my assumption correct ?
  If yes, how do I exactly do the rendering part (code example would be highly appreciated) ?