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• Merge commit ’4f7723cb3b913c577842a5bb088c804ddacac8df’

  17 novembre 2016, par Hendrik Leppkes

  Merge commit ’4f7723cb3b913c577842a5bb088c804ddacac8df’
  * commit ’4f7723cb3b913c577842a5bb088c804ddacac8df’ :
  
    movenc : Add an option for skipping writing the mfra/tfra/mfro trailer
  Merged-by : Hendrik Leppkes <h.leppkes@gmail.com>
  

  • [DH] libavformat/movenc.c
  • [DH] libavformat/movenc.h

• h264/aac encoded video not playing in Edge, but working Chrome and Firefox

  7 août 2016, par Flai

  I am trying to encode a video to use in the HTML5 video-Element. However, it doesn’t work in IE Edge on my PC. It does, however, work on :

  • Chrome, Firefox
  • My android smartphone (Chrome, Firefox)

  I have two very similar files (one being the big bucks bunny trailer, the second being the video I’m trying to encode) :

  Picture : Encoding-Settings

  As you can see "1.mp4" is playing correctly, and "3.mp4" is not.

  You can download the files here : 3.mp4 - change 3.mp4 to 1.mp4 to get the other video.

  My ffmpeg-commandline is this :

  ffmpeg.exe -i in.mp4 -c:v libx264 -movflags faststart -profile:v baseline -level 3.1 -preset slower -crf 30 -c:a copy 3.mp4

  However, I have also tried :

  ffmpeg.exe -i in.mp4 -c:v libx264 -profile:v baseline -level 3.1 -preset slower -crf 30 -c:a copy 3.mp4

  and

  ffmpeg.exe -i in.mp4 -c:v libx264 -preset slower -crf 30 -c:a copy 3.mp4

  The output of ffprobe of the result is (for the file that works) :

  $ ffprobe.exe 1.mp4
  
  ffprobe version N-81045-g450cf40 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.4.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 28.100 / 55. 28.100
  
    libavcodec     57. 50.100 / 57. 50.100
  
    libavformat    57. 42.100 / 57. 42.100
  
    libavdevice    57.  0.102 / 57.  0.102
  
    libavfilter     6. 47.100 /  6. 47.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '1.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf55.12.102
  
    Duration: 00:00:33.05, start: 0.000000, bitrate: 891 kb/s
  
      Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 512x288, 770 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 127 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler

  and for the file that doesn’t work :

  $ ffprobe.exe 3.mp4
  
  ffprobe version N-81045-g450cf40 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.4.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 28.100 / 55. 28.100
  
    libavcodec     57. 50.100 / 57. 50.100
  
    libavformat    57. 42.100 / 57. 42.100
  
    libavdevice    57.  0.102 / 57.  0.102
  
    libavfilter     6. 47.100 /  6. 47.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '3.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.42.100
  
    Duration: 00:00:13.10, start: 0.000000, bitrate: 2317 kb/s
  
      Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 2143 kb/s, 60 fps, 60 tbr, 15360 tbn, 120 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 163 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler

  Can anyone point me in the right direction what the difference between the two videos is and how I can fix my ffmpeg command-line to play "3.mp4" in IE Edge ?

• Connection reset by peer, ffmpeg

  18 août 2016, par Johnnylin

  I have tried several ways and done a lot of search. I just cannot figure out why this happens.

  This is the thread. I did almost the same thing.

  https://ffmpeg.org/pipermail/libav-user/2014-March/006356.html

  When you use ffmpeg command line together with ffserver, it works. But when you use sample code. It just does not work.

  What is missing ?

  EDIT

  Hi all,

  I took the muxing.c example and modified it in order to send a stream
  through a network socket. I only made few modifications :

  main function now looks like :

  int main()
  
  {
  
      AVOutputFormat *fmt;
  
      AVFormatContext *oc;
  
      AVStream *audio_st, *video_st;
  
      AVCodec *audio_codec, *video_codec;
  
      double audio_time, video_time;
  
      int flush, ret;
  
  
  
      /* Initialize libavcodec, and register all codecs and formats. */
  
      av_register_all();
  
      avformat_network_init();
  
  
  
      /* allocate the output media context */
  
      avformat_alloc_output_context2(&oc, NULL, "mpegts", NULL);
  
      if (!oc) {
  
          printf("Could not deduce output format from file extension: using
  
  MPEG.\n");
  
          avformat_alloc_output_context2(&oc, NULL, "mpegts", NULL);
  
      }
  
      if (!oc)
  
          return 1;
  
  
  
      fmt = oc->oformat;
  
      //fmt->video_codec = AV_CODEC_ID_MPEG2VIDEO;
  
      //fmt->audio_codec = AV_CODEC_ID_MP3;
  
  
  
      /* Add the audio and video streams using the default format codecs
  
       * and initialize the codecs. */
  
      video_st = NULL;
  
      audio_st = NULL;
  
  
  
      if (fmt->video_codec != AV_CODEC_ID_NONE)
  
          video_st = add_stream(oc, &video_codec, fmt->video_codec);
  
      if (fmt->audio_codec != AV_CODEC_ID_NONE)
  
          audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
  
  
  
      /* Now that all the parameters are set, we can open the audio and
  
       * video codecs and allocate the necessary encode buffers. */
  
      if (video_st)
  
          open_video(oc, video_codec, video_st);
  
      if (audio_st)
  
          open_audio(oc, audio_codec, audio_st);
  
  
  
     av_dump_format(oc, 0, "http://localhost:8090/feed1.ffm", 1);
  
  
  
      /* open the output file, if needed */
  
      if (!(fmt->flags & AVFMT_NOFILE)) {
  
          ret = avio_open(&oc->pb, "http://localhost:8090/feed1.ffm",
  
  AVIO_FLAG_WRITE);
  
          if (ret < 0) {
  
              fprintf(stderr, "Could not open '%s': %s\n", "
  
  http://localhost:8090/feed1.ffm",
  
                      av_err2str(ret));
  
              return 1;
  
          }
  
      }
  
  
  
      /* Write the stream header, if any. */
  
      ret = avformat_write_header(oc, NULL);
  
      if (ret < 0) {
  
          fprintf(stderr, "Error occurred when opening output file: %s\n",
  
                  av_err2str(ret));
  
          return 1;
  
      }
  
  
  
      flush = 0;
  
      while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
  
          /* Compute current audio and video time. */
  
          audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val *
  
  av_q2d(audio_st->time_base) : INFINITY;
  
          video_time = (video_st && !video_is_eof) ? video_st->pts.val *
  
  av_q2d(video_st->time_base) : INFINITY;
  
  
  
          if (!flush &&
  
              (!audio_st || audio_time >= STREAM_DURATION) &&
  
              (!video_st || video_time >= STREAM_DURATION)) {
  
              flush = 1;
  
          }
  
  
  
          /* write interleaved audio and video frames */
  
          if (audio_st && !audio_is_eof && audio_time <= video_time) {
  
              write_audio_frame(oc, audio_st, flush);
  
          } else if (video_st && !video_is_eof && video_time < audio_time) {
  
              write_video_frame(oc, video_st, flush);
  
          }
  
      }
  
  
  
      /* Write the trailer, if any. The trailer must be written before you
  
       * close the CodecContexts open when you wrote the header; otherwise
  
       * av_write_trailer() may try to use memory that was freed on
  
       * av_codec_close(). */
  
      av_write_trailer(oc);
  
  
  
      /* Close each codec. */
  
      if (video_st)
  
          close_video(oc, video_st);
  
      if (audio_st)
  
          close_audio(oc, audio_st);
  
  
  
      if (!(fmt->flags & AVFMT_NOFILE))
  
          /* Close the output file. */
  
          avio_close(oc->pb);
  
  
  
      /* free the stream */
  
      avformat_free_context(oc);
  
  
  
      return 0;
  
  }  

  and, in order to avoid a warning about channel layout not specified, I
  added :

  c->channel_layout = av_get_default_channel_layout(c->channels);
  
  
  
  in function AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
  
                          enum AVCodecID codec_id)

  just under the row c->channels = 2 ;

  I also raised a ffserver with the following configuration (showing only
  feed lines) :

  <feed>
  
          File /tmp/feed1.ffm
  
          FileMaxSize 1GB
  
          ACL allow 127.0.0.1
  
          ACL allow 192.168.0.0 192.168.255.255
  
  </feed>

  ffserver is working fine if I feed it with a ffmpeg commandline, e.g :

  ffmpeg -r 25 -i movie.mp4 -acodec libfdk_aac  -ab 128k -vcodec libx264 -fpre libx264-fast.ffpreset http://localhost:8090/feed1.ffm

  But with my example, I can write only few frames and after that may muxing
  modified program ends with :

  Error while writing video frame : Connection reset by peer

  I tried also different codecs (h264) and format (flv), turning out in a
  different number of frames written, but eventually I got the same error
  above.

  ffserver do not reports errors at all, only write:
  
  Tue Mar  4 12:55:10 2014 127.0.0.1 - - [POST] "/feed1.ffm HTTP/1.1" 200 4096
  
  confirming that the communication socket was open

  What am i missing ??

  Thanks