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• Fix —output-prefix with input-files in sub-directories

  29 May 2018, by orbea

  Fix —output-prefix with input-files in sub-directories
  And make sure to reserve the whole file path when not using —output-prefix.
  Fixes https://sourceforge.net/p/flac/bugs/463/
  
  Signed-off-by: Erik de Castro Lopo <erikd@mega-nerd.com>
  

  • [DH] src/flac/main.c

• Converting videos to flv using ffmpeg

  6 January 2012, by user703526

  In my c# application, i am writing code for converting any video format to flv format. For this FFMPEG is used.

  Some times an exceptions is occuring like:

  Attempted to read or write protected memory. This is often an indication that other memory is corrupt

  Below is my code from where the exception throwing,

      IntPtr pFormatContext;
  
      FFmpeg.av_register_all();
  
  
  
      int ret;
  
      ret = FFmpeg.av_open_input_file(out pFormatContext, this.Filename, IntPtr.Zero, 0, IntPtr.Zero);
  
  
  
      if (ret < 0)
  
      {
  
          Trace.WriteLine("couldn't open input file");
  
  
  
          FFmpeg.av_free_static();
  
          return;
  
      }
  
  
  
  
  
      try
  
      {
  
          ret = FFmpeg.av_find_stream_info(pFormatContext);
  
  
  
          if (ret < 0)
  
          {
  
              Trace.WriteLine("couldnt find stream informaion");
  
              FFmpeg.av_close_input_file(pFormatContext);
  
              FFmpeg.av_free_static();
  
              return;
  
          }
  
  
  
  
  
          FFmpeg.AVFormatContext formatContext = (FFmpeg.AVFormatContext)Marshal.PtrToStructure(pFormatContext, typeof(FFmpeg.AVFormatContext));
  
  
  
          Duration = formatContext.duration / FFmpeg.AV_TIME_BASE;
  
  
  
          for (int i = 0; i < formatContext.nb_streams; ++i)
  
          {
  
              FFmpeg.AVStream stream = (FFmpeg.AVStream)Marshal.PtrToStructure(formatContext.streams[i], typeof(FFmpeg.AVStream));
  
              FFmpeg.AVCodecContext codec = (FFmpeg.AVCodecContext)Marshal.PtrToStructure(stream.codec, typeof(FFmpeg.AVCodecContext));
  
  
  
              if (codec.codec_type == FFmpeg.CodecType.CODEC_TYPE_VIDEO)
  
              {
  
                  Height = codec.height;
  
                  Width = codec.width;
  
  
  
                          Type = FileType.flv;
  
                          MimeType = "video/x-flv";
  
  
  
              }
  
  
  
          }
  
      }
  
      catch (Exception ex)
  
      {
  
          Trace.WriteLine("FFMpeg failed to understand the file");
  
      }
  
  
  
      FFmpeg.av_close_input_file(pFormatContext);
  
      FFmpeg.av_free_static();
  
  }

  And from the above code this ret = FFmpeg.av_find_stream_info(pFormatContext); line throws memory corrupt exception.
  Please help me to solve this issue.

• Transcode HLS Segments individually using FFMPEG

  27 May 2013, by rayh

  I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

  Here is an example ffmpeg command line:

  ffmpeg -threads 1 -nostdin -loglevel verbose \
  
     -nostdin -y -i input.ts -c:a libfdk_aac \
  
     -ac 2 -b:a 64k -y -metadata -vn output.ts

  Inspecting an example sound file shows that there is a gap at the end of the audio:

  

  And the start of the file looks suspiciously attenuated (although this may not be an issue):

  

  My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

  Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream?

  ** UPDATE 1 **

  Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata?)

  
  

  ** UPDATED 2 **

  So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result:

  
  

  I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

  ** UPDATE 3 **

  According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

  For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.