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• FFMPEG conversion issue in PHP | 0KB conversion issue

  27 mai 2012, par PHPLearner

  i am using below code in the exec() command, it works in other server, in my dev region just converting into empty flv file.

  i have paid to my host to reinstall the ffmpeg, they have confirmed its good still not getting the output.

  they are claiming that i have a issue in the below syntax. does any one have any thoughts on this ?

  ffmpeg -i /home/hereitis/uploads/videos/MVI_0572.AVI -f flv -s 320x240 /home/hereitis/videos/flv/MVI_0572.flv 

  Thank you so mcuh for your interest to read this.

• Crop video by frame and save output as video

  3 avril 2018, par D_Corson

  I’m trying to resolve a question I had posted earlier (trying to only use FFMPEG) and am still stuck and hoping someone else has a solution.

  I have a video that I want to crop frame by frame with a varying locations to extract a region of interest. I can currently do this using moviepy which has been excellent, but I would like to try and solve this using only FFMPEG. The added constraints are that I would like to be able to crop the frame and put this in a new video without having to save the cropped images locally and compiling them.

  Thanks

• Increase/Decrease audio volume using FFmpeg

  9 juin 2016, par williamtroup

  I’m am currently using C# invokes to call the FFmpeg APIs to handle video and audio. I have the following code in place to extract the audio from a video and write it to a file.

  while (ffmpeg.av_read_frame(formatContext, &packet) >= 0)
  
  {
  
      if (packet.stream_index == streamIndex)
  
      {
  
          while (packet.size > 0)
  
          {
  
              int frameDecoded;
  
              int frameDecodedResult = ffmpeg.avcodec_decode_audio4(codecContext, frame, &frameDecoded, packet);
  
  
  
              if (frameDecoded > 0 && frameDecodedResult >= 0)
  
              {
  
                  //writeAudio.WriteFrame(frame);
  
  
  
                  packet.data += totalBytesDecoded;
  
                  packet.size -= totalBytesDecoded;
  
              }
  
          }
  
  
  
          frameIndex++;
  
      }
  
  
  
      Avcodec.av_free_packet(&packet);
  
  }

  This is all working correctly. I’m currently using the FFmpeg.AutoGen project for the API access.

  I want to be able to increase/decrease the volume of the audio before its written to the file, but I cannot seem to find a command or any help with this. Does it have to be done manually ?

  Update 1 :

  After receiving some help, this is the class layout I have :

  public unsafe class FilterVolume
  
  {
  
      #region Private Member Variables
  
  
  
      private AVFilterGraph* m_filterGraph = null;
  
      private AVFilterContext* m_aBufferSourceFilterContext = null;
  
      private AVFilterContext* m_aBufferSinkFilterContext = null;
  
  
  
      #endregion
  
  
  
      #region Private Constant Member Variables
  
  
  
      private const int EAGAIN = 11;
  
  
  
      #endregion
  
  
  
      public FilterVolume(AVCodecContext* codecContext, AVStream* stream, float volume)
  
      {
  
          CodecContext = codecContext;
  
          Stream = stream;
  
          Volume = volume;
  
  
  
          Initialise();
  
      }
  
  
  
      public AVFrame* Adjust(AVFrame* frame)
  
      {
  
          AVFrame* returnFilteredFrame = ffmpeg.av_frame_alloc();
  
  
  
          if (m_aBufferSourceFilterContext != null && m_aBufferSinkFilterContext != null)
  
          {
  
              int bufferSourceAddFrameResult = ffmpeg.av_buffersrc_add_frame(m_aBufferSourceFilterContext, frame);
  
              if (bufferSourceAddFrameResult < 0)
  
              {
  
              }
  
  
  
              int bufferSinkGetFrameResult = ffmpeg.av_buffersink_get_frame(m_aBufferSinkFilterContext, returnFilteredFrame);
  
              if (bufferSinkGetFrameResult < 0 && bufferSinkGetFrameResult != -EAGAIN)
  
              {
  
              }
  
          }
  
  
  
          return returnFilteredFrame;
  
      }
  
  
  
      public void Dispose()
  
      {
  
          Cleanup(m_filterGraph);
  
      }
  
  
  
      #region Private Properties
  
  
  
      private AVCodecContext* CodecContext { get; set; }
  
      private AVStream* Stream { get; set; }
  
      private float Volume { get; set; }
  
  
  
      #endregion
  
  
  
      #region Private Setup Helper Functions
  
  
  
      private void Initialise()
  
      {
  
          m_filterGraph = GetAllocatedFilterGraph();
  
  
  
          string aBufferFilterArguments = string.Format("sample_fmt={0}:channel_layout={1}:sample_rate={2}:time_base={3}/{4}",
  
              (int)CodecContext->sample_fmt,
  
              CodecContext->channel_layout,
  
              CodecContext->sample_rate,
  
              Stream->time_base.num,
  
              Stream->time_base.den);
  
  
  
          AVFilterContext* aBufferSourceFilterContext = CreateFilter("abuffer", m_filterGraph, aBufferFilterArguments);
  
          AVFilterContext* volumeFilterContext = CreateFilter("volume", m_filterGraph, string.Format("volume={0}", Volume));
  
          AVFilterContext* aBufferSinkFilterContext = CreateFilter("abuffersink", m_filterGraph);
  
  
  
          LinkFilter(aBufferSourceFilterContext, volumeFilterContext);
  
          LinkFilter(volumeFilterContext, aBufferSinkFilterContext);
  
  
  
          SetFilterGraphConfiguration(m_filterGraph, null);
  
  
  
          m_aBufferSourceFilterContext = aBufferSourceFilterContext;
  
          m_aBufferSinkFilterContext = aBufferSinkFilterContext;
  
      }
  
  
  
      #endregion
  
  
  
      #region Private Cleanup Helper Functions
  
  
  
      private static void Cleanup(AVFilterGraph* filterGraph)
  
      {
  
          if (filterGraph != null)
  
          {
  
              ffmpeg.avfilter_graph_free(&filterGraph);
  
          }
  
      }
  
  
  
      #endregion
  
  
  
      #region Provate Helpers
  
  
  
      private AVFilterGraph* GetAllocatedFilterGraph()
  
      {
  
          AVFilterGraph* filterGraph = ffmpeg.avfilter_graph_alloc();
  
          if (filterGraph == null)
  
          {
  
          }
  
  
  
          return filterGraph;
  
      }
  
  
  
      private AVFilter* GetFilterByName(string name)
  
      {
  
          AVFilter* filter = ffmpeg.avfilter_get_by_name(name);
  
          if (filter == null)
  
          {
  
          }
  
  
  
          return filter;
  
      }
  
  
  
      private void SetFilterGraphConfiguration(AVFilterGraph* filterGraph, void* logContext)
  
      {
  
          int filterGraphConfigResult = ffmpeg.avfilter_graph_config(filterGraph, logContext);
  
          if (filterGraphConfigResult < 0)
  
          {
  
          }
  
      }
  
  
  
      private AVFilterContext* CreateFilter(string filterName, AVFilterGraph* filterGraph, string filterArguments = null)
  
      {
  
          AVFilter* filter = GetFilterByName(filterName);
  
          AVFilterContext* filterContext;
  
  
  
          int aBufferFilterCreateResult = ffmpeg.avfilter_graph_create_filter(&filterContext, filter, filterName, filterArguments, null, filterGraph);
  
          if (aBufferFilterCreateResult < 0)
  
          {
  
          }
  
  
  
          return filterContext;
  
      }
  
  
  
      private void LinkFilter(AVFilterContext* source, AVFilterContext* destination)
  
      {
  
          int filterLinkResult = ffmpeg.avfilter_link(source, 0, destination, 0);
  
          if (filterLinkResult < 0)
  
          {
  
          }
  
      }
  
  
  
      #endregion
  
  }

  The Adjust() function is called after a frame is decoded. I’m currently getting a -22 error when av_buffersrc_add_frame() is called. This indicates that a parameter is invalid, but after debugging, I cannot see anything that would be causing this.

  This is how the code is called :

  while (ffmpeg.av_read_frame(formatContext, &packet) >= 0)
  
  {
  
      if (packet.stream_index == streamIndex)
  
      {
  
          while (packet.size > 0)
  
          {
  
              int frameDecoded;
  
              int frameDecodedResult = ffmpeg.avcodec_decode_audio4(codecContext, frame, &frameDecoded, packet);
  
  
  
              if (frameDecoded > 0 && frameDecodedResult >= 0)
  
              {
  
                  AVFrame* filteredFrame = m_filterVolume.Adjust(frame);
  
  
  
                  //writeAudio.WriteFrame(filteredFrame);
  
  
  
                  packet.data += totalBytesDecoded;
  
                  packet.size -= totalBytesDecoded;
  
              }
  
          }
  
  
  
          frameIndex++;
  
      }
  
  
  
      Avcodec.av_free_packet(&packet);
  
  }

  Update 2 :

  Cracked it. The "channel_layout" option in the filter argument string is supposed to be a hexadecimal. This is what the string formatting should look like :

  string aBufferFilterArguments = string.Format("sample_fmt={0}:channel_layout=0x{1}:sample_rate={2}:time_base={3}/{4}",
  
      (int)CodecContext->sample_fmt,
  
      CodecContext->channel_layout,
  
      CodecContext->sample_rate,
  
      Stream->time_base.num,
  
      Stream->time_base.den);