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• ffmpeg removing silence makes mp3 longer ?

  13 août 2017, par pocketg99

  I’ve been using the following command to attempt to remove silent segments from an mp3 file

  ffmpeg -i "podcasts/audio1.mp3" -af silenceremove=1:0:-50dB "/tmp/pod-sil.mp3"

  For some reason the resulting mp3 is twice as log as the input mp3. It is not half as fast. There does not appear to be any duplicated audio. There is some silence, but not an hour’s worth. For a given portion of the input file, you can find the same thing in the output file by going to twice the timestamp of the input file.

  The files are long so I have not yet listened to them all the way through. I really have no idea where the extra length is coming from, the files seem normal.

  Here is the full output from ffmpeg

  ffmpeg version 2.8.11-0ubuntu0.16.04.1 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0x21880e0] Skipping 0 bytes of junk at 0. 
  
  [mp3 @ 0x21880e0] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from 'podcasts/audio1.mp3':
  
    Duration: 01:00:00.20, start: 0.000000, bitrate: 320 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
  File '/tmp/pod-sil.mp3' already exists. Overwrite ? [y/N] y
  
  Output #0, mp3, to '/tmp/pod-sil.mp3':
  
    Metadata:
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0x21999e0] Trying to remove 1152 samples, but the queue is empty
  
  size=   56253kB time=01:00:00.16 bitrate= 128.0kbits/s    
  
  video:0kB audio:56253kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000439%

• dfa : Disallow odd width/height and add proper bounds check for DDS1 chunks

  11 août 2017, par Diego Biurrun

  dfa : Disallow odd width/height and add proper bounds check for DDS1 chunks
  DDS1 chunks are decoded in 2x2 blocks, odd chunk width or height is not
  
  allowed in that case. Also ensure that the decode buffer is big enough
  
  for all blocks being processed.
  Bug-Id : CVE-2017-9992
  
  CC : libav-stable@libav.org
  

  • [DBH] libavcodec/dfa.c

• ffmpeg, 'protocol not found' error

  15 août 2017, par gogoer

  i’m using next code :

  const char *sFileOutput;
  
  AVOutputFormat *ofmt;
  
  AVFormatContext *ofcx;
  
  
  
          int main( int argc, char* argv[] )
  
          {
  
              av_log_set_level( AV_LOG_DEBUG );
  
              av_register_all();
  
              avcodec_register_all();
  
              avformat_network_init();
  
  
  
              char s1[40]={0};
  
              const time_t timer = time(NULL);
  
              u = localtime(&timer);
  
              strftime(s1, 40, "%d.%m.%Y-%H:%M:%S.avi", u);
  
              sFileOutput=&s1;
  
  
  
              //char *sFileOutput = "01.01.2017-23.23.23.avi";
  
  
  
              ofmt = av_guess_format( NULL, sFileOutput, NULL );
  
              ofcx = avformat_alloc_context();
  
              ofcx->oformat = ofmt;
  
              int ret2=avio_open( &ofcx->pb, sFileOutput, AVIO_FLAG_WRITE);
  
              if(ret2<0){
  
                  fprintf(stderr, "\nError occurred when opening output file: %s\n",av_err2str(ret2));
  
              }
  
          }

  When i run it, i have error in console :

  Error occurred when opening output file: Protocol not found

  but if i uncomment string

  char *sFileOutput = "01.01.2017-23.23.23.avi";

  evirything is ok, progam is working without errors. please tell me what is wrong.