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• FFmpeg changing Keyframe Interval / GoP values while reencoding a Livestream ?

  17 juin 2015, par eCronik

  I am using ffmpeg in an nginx environment to reencode Livestreams on Ubuntu.

  So there are 2 Livestreams with the exact identical options going into the reencode. Both are streaming to a server in germany - one comes from germany, the other comes from the US eastcoast. So I have the problem, that the receiver after the reencode moans after some time about the Keyframe Interval growing higher of the stream coming from the US - until a certain point where it stops. This is not always happening in the same amount of time or to the same extent.

  This is the line I am using to reencode :

  exec_push ffmpeg -i rtmp://localhost/input/$name -c:v libx264 -preset slow -profile:v main -b:v 2175k -minrate 2175k -maxrate 2175k -bufsize 2175k -r 40 -g 80 -c:a aac -strict -2 -b:a 160k -f flv rtmp://server \-c copy -f flv rtmp://server;

  Does anybody have a clue what could cause this and how to prevent it ? Changing x264 cpu preset on the machine where the stream comes from to something faster doesn’t work, and the reencode-server is just at 50% usage with ’slow’ as well.

  Here my version info :

  ffmpeg version 2.6.3 Copyright (c) 2000-2015 the FFmpeg developers
  
  built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvo-aacenc --enable-libvidstab
  
  libavutil      54. 20.100 / 54. 20.100
  
  libavcodec     56. 26.100 / 56. 26.100
  
  libavformat    56. 25.101 / 56. 25.101
  
  libavdevice    56.  4.100 / 56.  4.100
  
  libavfilter     5. 11.102 /  5. 11.102
  
  libavresample   2.  1.  0 /  2.  1.  0
  
  libswscale      3.  1.101 /  3.  1.101
  
  libswresample   1.  1.100 /  1.  1.100
  
  libpostproc    53.  3.100 / 53.  3.100

  Thanks a bunch,
  eC

• Not able to decode mp4 file using latest ffmpeg library : av_decode_video2

  8 février 2017, par suvirai

  I am writing a wrapper code around latest ffmpeg library. I am supplying MP4 files from local system. My problem is that I am unable to get any decoded frames when I use av_decode_video2(). The return value comes out to be negative. I have used av_read_frame() which returns 0. I googled about the problem I am facing but no where could I find the correct explanation. Please give me insight here. Pasting the pseudo code here.

      av_init_packet(avpkt);
  
      picture=av_frame_alloc();
  
      pFrameRGB=av_frame_alloc();
  
      codec = avcodec_find_decoder(CODEC_ID_H264);
  
      c= avcodec_alloc_context3(codec)
  
      avcodec_open2(decoderLibraryData->c, decoderLibraryData->codec, NULL)
  
      FormatContext = avformat_alloc_context();
  
      char *pUrl ="./1.MP4";
  
  
  
      iRet = avformat_open_input(atContext, pUrl, pFmt, NULL);
  
  
  
      if(FormatContext == NULL)
  
      {
  
          printf("could not assign any memory !!!!!!!!! \n");
  
      }
  
  
  
      avformat_find_stream_info(FormatContext, NULL);
  
  
  
  
  
      while(av_read_frame(FormatContext,avpkt) >= 0)
  
      {
  
  
  
        len = avcodec_decode_video2(c, picture, &got_picture,avpkt);
  
  
  
        printf("CODEC MANAGER len %d Frame decompressed %d \n",len,got_picture);
  
  
  
        if (len <= 0) 
  
        {
  
          return ERROR;
  
        }
  
      }
  
  }
  
  
  
  
  
  
  
          if(lastHeight != 0 && lastWidth != 0)
  
          {
  
              if(lastWidth != c->width || lastHeight != c->height )
  
              {
  
                  av_free(buffer);
  
                  buffer = NULL;
  
                  lastWidth = c->width;
  
                  lastHeight = c->height;
  
  
  
              }
  
          }
  
          else
  
          {
  
              lastWidth = c->width;
  
              lastHeight = c->height;
  
          }
  
          decodeFlag = 1;
  
          if(!buffer)
  
          {
  
              int numBytes;
  
              v_mutex_lock(globalCodecLock);
  
              switch(inPixFormat)
  
              {
  
              case RGB:
  
  
  
  
  
                  // Determine required buffer size and allocate buffer
  
                  numBytes=avpicture_get_size(PIX_FMT_RGB24, c->width, c->height);
  
  
  
                  buffer=(uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
  
                  avpicture_fill((AVPicture *)pFrameRGB,buffer,PIX_FMT_RGB24,c->width, c->height);
  
  
  
                  if(cntxt)
  
                      sws_freeContext(cntxt);
  
  
  
                  cntxt = sws_getContext(c->width, c->height, c->pix_fmt,
  
                          c->width, c->height, PIX_FMT_RGB24, SWS_BICUBIC, NULL, NULL, NULL);
  
  
  
                  break;
  
  
  
              }
  
              v_mutex_unlock(globalCodecLock);
  
              if(cntxt == NULL)
  
              {
  
                  printf("sws_getContext error\n");
  
                  return ERROR;
  
              }
  
              }
  
  
  
          {
  
              sws_scale(cntxt, picture->data, picture->linesize, 0, c->height, pFrameRGB->data, pFrameRGB->linesize);
  
              if(rgbBuff)
  
              {
  
  
  
  
  
                  if(c->width <= *width && c->height <= *height)
  
                  {                   
  
                      saveFrame(pFrameRGB, c->width, c->height, rgbBuff,inPixFormat);
  
  
  
                      *width = c->width;
  
                      *height = c->height;
  
                      rs = SUCCESS;
  
                      break;
  
                  }
  
                  else
  
                  {
  
                      rs = VA_LOWBUFFERSIZE;
  
                  }
  
              }
  
              else
  
              {
  
                  rs = VA_LOWBUFFERSIZE;
  
              }
  
          }
  
          if(width)
  
          {
  
              *width = c->width;
  
          }
  
          if(height)
  
          {
  
              *height = c->height;
  
          }
  
          if(rs == VA_LOWBUFFERSIZE)
  
          {
  
              break;
  
          }

  I am getting the return value of av_read_frame as 0 but av_decode_video2 returns value in negative. I am not able to get any clue here.

• "MPEG audio header not found" error when opening with TagLib after converting with ffmpeg

  6 juillet 2015, par Avrohom Yisroel

  I converted a wma file simply by doing something like this...

  ffmpeg -i song.wma -f mp3 song.mp3

  I can then play the mp3 file in Windows Media player, so it looks like the conversion worked.

  However, if I try to open the file in TagLib, I get an error "MPEG audio header not found" on the following line...

  TagLib.File tf = TagLib.File.Create("song.mp3");

  I’ve tried this on a few wma files, so it’s not just that one that’s at fault.

  Anyone have any idea what I did wrong ? I find the docs for ffmpeg pretty overwhelming, and as a complete ignoramus in the field of audio encoding, I haven’t a clue what most of it means. Could be I’m missing something in the conversion, although that wouldn’t explain why WMP can play it but TagLib can’t open it.