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• FFMPEG : Decode only portion of a frame

  17 avril 2019, par SAMPro

  I am using libav libraries to decode video and process frames in order to do some image processing. However I have limited resource to decode frames and no hardware acceleration at all. So decoding is the critical section in my project (regarding speed).

  Because only some section of frame is needed to processed, I’m wondering is it possible to decode only a region of interest (a rectangle) and the rest of the frame is not decoded ? And does this make the decoding faster ?

  Edit :
  I have decoding H.264 codec.

• 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.

• 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);