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• FFmpeg encoding live audio to aac issue

  12 juillet 2015, par Ruurd Adema

  I’m trying to encode live raw audio coming from a Blackmagic Decklink input card to a mov file with AAC encoding.

  The issue is that the audio sounds distorted and plays to fast.

  I created the software based on a couple of examples/tutorials including the Dranger tutorial and examples on Github (and of course the examples in the FFmpeg codebase).

  Honestly, at this moment I don’t exactly know what the cause of the problem is. I’m thinking about PTS/DTS values or a timebase mismatch (because of the too fast playout), I tried a lot of things, including working with an av_audio_fifo.

  • When outputting to the mov file with the AV_CODEC_ID_PCM_S16LE codec, everything works well
  • When outputting to the mov file with the AV_CODEC_ID_AAC codec, the problems occur
  • When writing RAW audio VLC media info shows :
    Type : Audio, Codec : PCM S16 LE (sowt), Language : English, Channels : Stereo, Sample rate : 48000 Hz, Bits per sample.
  • When writing with AAC codec VLC media info shows :
    Type : Audio, Codec : MPEG AAC Audio (mp4a), Language : English, Channels : Stereo, Sample rate : 48000 Hz.

  Any idea(s) of what’s causing the problems ?

  Code

  // Create output context
  
  
  
  output_filename = "/root/movies/encoder_debug.mov";
  
  output_format_name = "mov";
  
  if (avformat_alloc_output_context2(&output_fmt_ctx, NULL, output_format_name, output_filename) < 0)
  
  {
  
      printf("[ERROR] Unable to allocate output format context for output: %s\n", output_filename);
  
  } 
  
  
  
  // Create audio output stream
  
  
  
  static AVStream *encoder_add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
  
  {
  
      AVCodecContext  *c;
  
      AVCodec         *codec;
  
      AVStream        *st;
  
  
  
      st = avformat_new_stream(oc, NULL);
  
      if (!st) 
  
      {
  
          printf("[ERROR] Could not allocate new audio stream!\n");
  
          exit(-1);
  
      }
  
  
  
      c                 = st->codec;
  
      c->codec_id       = codec_id;
  
      c->codec_type     = AVMEDIA_TYPE_AUDIO;
  
      c->sample_fmt     = AV_SAMPLE_FMT_S16;
  
      c->sample_rate    = decklink_config()->audio_samplerate;
  
      c->channels       = decklink_config()->audio_channel_count;
  
      c->channel_layout = av_get_default_channel_layout(decklink_config()->audio_channel_count);
  
      c->time_base.den  = decklink_config()->audio_samplerate;
  
      c->time_base.num  = 1;
  
  
  
      if (codec_id == AV_CODEC_ID_AAC)
  
      {
  
          c->bit_rate       = 96000;  
  
          //c->profile = FF_PROFILE_AAC_MAIN; //FIXME Generates error: "Unable to set the AOT 1: Invalid config"
  
          // Allow the use of the experimental AAC encoder
  
          c->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; 
  
      }
  
  
  
      // Some formats want stream headers to be seperate (global?)
  
      if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
      {
  
          c->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
      }
  
  
  
      codec = avcodec_find_encoder(c->codec_id);
  
      if (!codec)
  
      {
  
          printf("[ERROR] Audio codec not found\n");
  
          exit(-1);
  
      }
  
  
  
      if (avcodec_open2(c, codec, NULL) < 0)
  
      {
  
          printf("[ERROR] Could not open audio codec\n");
  
          exit(-1);
  
      }
  
  
  
      return st;
  
  }
  
  
  
  // En then, at every incoming frame this function gets called:
  
  
  
  void encoder_handle_incoming_frame(IDeckLinkVideoInputFrame *videoframe, IDeckLinkAudioInputPacket *audiopacket)
  
  {
  
      void *pixels = NULL;
  
      int   pitch = 0;
  
      int got_packet = 0;
  
  
  
      void *audiopacket_data          = NULL;
  
      long  audiopacket_sample_count  = 0;
  
      long  audiopacket_size          = 0;
  
      long  audiopacket_channel_count = 2;
  
  
  
      if (audiopacket)
  
      {  
  
          AVPacket pkt = {0,0,0,0,0,0,0,0,0,0,0,0,0,0};
  
          AVFrame *frame;
  
          BMDTimeValue audio_pts;
  
          int requested_size;
  
          static int last_pts1, last_pts2 = 0;
  
  
  
          audiopacket_sample_count  = audiopacket->GetSampleFrameCount();
  
          audiopacket_channel_count = decklink_config()->audio_channel_count;
  
          audiopacket_size          = audiopacket_sample_count * (decklink_config()->audio_sampletype/8) * audiopacket_channel_count;
  
  
  
          audiopacket->GetBytes(&audiopacket_data);
  
  
  
          av_init_packet(&pkt);    
  
  
  
          printf("\n=== Audiopacket: %d ===\n", audio_stream->codec->frame_number);
  
  
  
          if (AUDIO_TYPE == AV_CODEC_ID_PCM_S16LE)
  
          {
  
              audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
  
  
  
              pkt.pts          = audio_pts;
  
              pkt.dts          = pkt.pts; 
  
              pkt.flags       |= AV_PKT_FLAG_KEY;                 // TODO: Make sure if this still applies
  
              pkt.stream_index = audio_stream->index;
  
              pkt.data         = (uint8_t *)audiopacket_data;
  
              pkt.size         = audiopacket_size;
  
  
  
              printf("[PACKET] size:              %d\n", pkt.size);
  
              printf("[PACKET] pts:               %li\n", pkt.pts);
  
              printf("[PACKET] pts delta:         %li\n", pkt.pts - last_pts2);
  
              printf("[PACKET] duration:          %d\n", pkt.duration);
  
              last_pts2 = pkt.pts;
  
  
  
              av_interleaved_write_frame(output_fmt_ctx, &pkt);
  
          } 
  
          else if (AUDIO_TYPE == AV_CODEC_ID_AAC)
  
          {
  
              frame = av_frame_alloc();
  
              frame->format = audio_stream->codec->sample_fmt;
  
              frame->channel_layout = audio_stream->codec->channel_layout;
  
              frame->sample_rate = audio_stream->codec->sample_rate;
  
              frame->nb_samples = audiopacket_sample_count;
  
  
  
              requested_size = av_samples_get_buffer_size(NULL, audio_stream->codec->channels, audio_stream->codec->frame_size, audio_stream->codec->sample_fmt, 1);
  
  
  
              audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
  
  
  
              printf("[DEBUG] Sample format:      %d\n", frame->format);
  
              printf("[DEBUG] Channel layout:     %li\n", frame->channel_layout);
  
              printf("[DEBUG] Sample rate:        %d\n", frame->sample_rate);
  
              printf("[DEBUG] NB Samples:         %d\n", frame->nb_samples);
  
              printf("[DEBUG] Datasize:           %li\n", audiopacket_size);
  
              printf("[DEBUG] Requested datasize: %d\n", requested_size);
  
              printf("[DEBUG] Too less/much:      %li\n", audiopacket_size - requested_size);
  
              printf("[DEBUG] Framesize:          %d\n", audio_stream->codec->frame_size);
  
              printf("[DEBUG] Audio pts:          %li\n", audio_pts);
  
              printf("[DEBUG] Audio pts delta:    %li\n", audio_pts - last_pts1);
  
              last_pts1 = audio_pts;
  
  
  
              frame->pts = audio_pts;
  
  
  
              if (avcodec_fill_audio_frame(frame, audiopacket_channel_count, audio_stream->codec->sample_fmt, (const uint8_t *)audiopacket_data, audiopacket_size, 0) < 0)
  
              {
  
                  printf("[ERROR] Filling audioframe failed!\n");
  
                  exit(-1);
  
              }
  
  
  
              got_packet = 0;
  
              if (avcodec_encode_audio2(audio_stream->codec, &pkt, frame, &got_packet) != 0)
  
              {
  
                  printf("[ERROR] Encoding audio failed\n");
  
              }
  
  
  
              if (got_packet) 
  
              {
  
                  pkt.stream_index = audio_stream->index;
  
                  pkt.flags       |= AV_PKT_FLAG_KEY; 
  
  
  
                  //printf("[PACKET] size:              %d\n", pkt.size);
  
                  //printf("[PACKET] pts:               %li\n", pkt.pts);
  
                  //printf("[PACKET] pts delta:         %li\n", pkt.pts - last_pts2);
  
                  //printf("[PACKET] duration:          %d\n", pkt.duration);
  
                  //printf("[PACKET] timebase codec:    %d/%d\n", audio_stream->codec->time_base.num, audio_stream->codec->time_base.den);
  
                  //printf("[PACKET] timebase stream:   %d/%d\n", audio_stream->time_base.num, audio_stream->time_base.den);
  
                  last_pts2 = pkt.pts;
  
  
  
                  av_interleaved_write_frame(output_fmt_ctx, &pkt);
  
              }
  
              av_frame_free(&frame); 
  
          }
  
  
  
          av_free_packet(&pkt);
  
      }
  
      else
  
      {
  
          printf("[WARNING] No audiopacket received!\n");
  
      }
  
  
  
      static int count = 0;
  
      count++;
  
  }

• FFmpeg - downmixing FLAC 6.1 to AAC 5.1

  7 juillet 2014, par Martijn

  I can’t seem to figure out how to do this. I’ve been staring at these commands :
  https://trac.ffmpeg.org/wiki/AudioChannelManipulation

  But to no avail. It’s a tad above my level, sadly. Here’s the ffmpeg -i output for the video in question :

  ffmpeg version N-64012-g61df081 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Jun 16 2014 22:01:59 with gcc 4.8.3 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --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-libsoxr --enable-libspeex--enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
  
    libavutil      52. 89.100 / 52. 89.100
  
    libavcodec     55. 67.100 / 55. 67.100
  
    libavformat    55. 43.100 / 55. 43.100
  
    libavdevice    55. 13.101 / 55. 13.101
  
    libavfilter     4.  8.100 /  4.  8.100
  
    libswscale      2.  6.100 /  2.  6.100
  
    libswresample   0. 19.100 /  0. 19.100
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Input #0, matroska,webm, from '[Coalgirls]_Spirited_Away_(1920x1038_Blu-ray_FLAC)_[92372194].mkv':
  
    Metadata:
  
      title           : Spirited Away
  
      encoder         : libebml v1.3.0 + libmatroska v1.4.0
  
      creation_time   : 2014-07-03 01:32:13
  
    Duration: 02:04:32.29, start: 0.000000, bitrate: 15972 kb/s
  
      Chapter #0.0: start 0.000000, end 99.099000
  
      Metadata:
  
        title           : 00:00:00.000
  
      Chapter #0.1: start 99.099000, end 196.238000
  
      Metadata:
  
        title           : 00:01:39.099
  
      Chapter #0.2: start 196.238000, end 443.526000
  
      Metadata:
  
        title           : 00:03:16.238
  
      Chapter #0.3: start 443.526000, end 645.395000
  
      Metadata:
  
        title           : 00:07:23.526
  
      Chapter #0.4: start 645.395000, end 1023.022000
  
      Metadata:
  
        title           : 00:10:45.395
  
      Chapter #0.5: start 1023.022000, end 1368.534000
  
      Metadata:
  
        title           : 00:17:03.022
  
      Chapter #0.6: start 1368.534000, end 1716.048000
  
      Metadata:
  
        title           : 00:22:48.534
  
      Chapter #0.7: start 1716.048000, end 2008.173000
  
      Metadata:
  
        title           : 00:28:36.048
  
      Chapter #0.8: start 2008.173000, end 2301.674000
  
      Metadata:
  
        title           : 00:33:28.173
  
      Chapter #0.9: start 2301.674000, end 2651.816000
  
      Metadata:
  
        title           : 00:38:21.674
  
      Chapter #0.10: start 2651.816000, end 2906.821000
  
      Metadata:
  
        title           : 00:44:11.816
  
      Chapter #0.11: start 2906.821000, end 3271.351000
  
      Metadata:
  
        title           : 00:48:26.821
  
      Chapter #0.12: start 3271.351000, end 3729.017000
  
      Metadata:
  
        title           : 00:54:31.351
  
      Chapter #0.13: start 3729.017000, end 4091.587000
  
      Metadata:
  
        title           : 01:02:09.017
  
      Chapter #0.14: start 4091.587000, end 4476.847000
  
      Metadata:
  
        title           : 01:08:11.587
  
      Chapter #0.15: start 4476.847000, end 4750.579000
  
      Metadata:
  
        title           : 01:14:36.847
  
      Chapter #0.16: start 4750.579000, end 5139.760000
  
      Metadata:
  
        title           : 01:19:10.579
  
      Chapter #0.17: start 5139.760000, end 5478.890000
  
      Metadata:
  
        title           : 01:25:39.760
  
      Chapter #0.18: start 5478.890000, end 5853.806000
  
      Metadata:
  
        title           : 01:31:18.890
  
      Chapter #0.19: start 5853.806000, end 6318.937000
  
      Metadata:
  
        title           : 01:37:33.806
  
      Chapter #0.20: start 6318.937000, end 6625.118000
  
      Metadata:
  
        title           : 01:45:18.937
  
      Chapter #0.21: start 6625.118000, end 6771.098000
  
      Metadata:
  
        title           : 01:50:25.118
  
      Chapter #0.22: start 6771.098000, end 6914.199000
  
      Metadata:
  
        title           : 01:52:51.098
  
      Chapter #0.23: start 6914.199000, end 7253.580000
  
      Metadata:
  
        title           : 01:55:14.199
  
      Chapter #0.24: start 7253.580000, end 7472.288000
  
      Metadata:
  
        title           : 02:00:53.580
  
      Stream #0:0: Video: h264 (High 10), yuv420p10le(tv, bt709), 1920x1038, SAR 1:1 DAR 320:173, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
  
      Metadata:
  
        title           : Spirited Away
  
      Stream #0:1(jpn): Audio: flac, 48000 Hz, 6.1, s32 (default)
  
      Metadata:
  
        title           : 6.1 FLAC
  
      Stream #0:2(eng): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Metadata:
  
        title           : 5.1 AC3
  
      Stream #0:3(fre): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Metadata:
  
        title           : 5.1 AC3
  
      Stream #0:4(ger): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Metadata:
  
        title           : 5.1 AC3
  
      Stream #0:5(fin): Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        title           : 2.0 AC3
  
      Stream #0:6(kor): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Metadata:
  
        title           : 5.1 AC3
  
      Stream #0:7(chi): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Metadata:
  
        title           : 5.1 AC3
  
      Stream #0:8(chi): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Metadata:
  
        title           : 5.1 AC3
  
      Stream #0:9(eng): Subtitle: ssa (default)
  
      Metadata:
  
        title           : English
  
      Stream #0:10(fre): Subtitle: ssa
  
      Metadata:
  
        title           : French
  
      Stream #0:11(ger): Subtitle: ssa
  
      Metadata:
  
        title           : German
  
      Stream #0:12(eng): Subtitle: ssa
  
      Metadata:
  
        title           : Songs + Signs
  
      Stream #0:13: Attachment: ttf
  
      Metadata:
  
        filename        : MyriadPro-Regular.otf
  
        mimetype        : application/x-truetype-font
  
      Stream #0:14: Attachment: ttf
  
      Metadata:
  
        filename        : MyriadPro-SemiboldIt.otf
  
        mimetype        : application/x-truetype-font
  
      Stream #0:15: Attachment: ttf
  
      Metadata:
  
        filename        : Vesta-Bold.otf
  
        mimetype        : application/x-truetype-font
  
      Stream #0:16: Attachment: ttf
  
      Metadata:
  
        filename        : Vesta-Bold_2.otf
  
        mimetype        : application/x-truetype-font
  
      Stream #0:17: Attachment: ttf
  
      Metadata:
  
        filename        : AR CENA_0.TTF
  
        mimetype        : application/x-truetype-font
  
      Stream #0:18: Attachment: ttf
  
      Metadata:
  
        filename        : tahomabd.ttf
  
        mimetype        : application/x-truetype-font
  
      Stream #0:19: Attachment: ttf
  
      Metadata:
  
        filename        : palai.ttf
  
        mimetype        : application/x-truetype-font
  
      Stream #0:20: Attachment: ttf
  
      Metadata:
  
        filename        : pala.ttf
  
        mimetype        : application/x-truetype-font

  As you can see, one of the streams is a FLAC 6.1 stream. I wanted to convert that to AAC, and I know how to do that, basically like this :

  ffmpeg -i "input.mkv" -codec:v copy -codec:a aac -strict -2 -b:a 320k -f matroska "output.mkv"

  But apparently AAC doesn’t support 6.1 audio :

  ...
  
  [aac @ 03b26860] Unsupported number of channels: 7
  
  Output #0, matroska, to 'd:\Movies\[Coalgirls]_Spirited_Away_(1920x1038_Blu-ray_FLAC)_[92372194].aac.mkv':
  
      Stream #0:0(jpn): Video: h264, yuv420p10le, 1920x1038 [SAR 1:1 DAR 320:173], q=2-31, 23.98 fps, 90k tbn, 1k tbc (default)
  
      Stream #0:1(jpn): Audio: aac, 0 channels, 128 kb/s (default)
  
      Metadata:
  
        encoder         : Lavc55.67.100 aac
  
      Stream #0:2(eng): Subtitle: ssa, 128 kb/s (default)
  
      Metadata:
  
        encoder         : Lavc55.67.100 ssa
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #0:1 -> #0:1 (flac (native) -> aac (aac))
  
    Stream #0:9 -> #0:2 (ssa (native) -> ssa (native))
  
  Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height

  That’s fine, so I wanted to downmix it to 5.1 and encode as AAC. But I can’t seem to work out how to. Any advice ?

• FFMPEG creates incorrect Source duration

  4 août 2016, par Byron Whitlock

  When transcoding movies from an AVI to mp4 sometimes FFMPEG sets the "Source Duration" incorrectly.

  This messes up playback on IOS devices. Specifically, it causes the video to cut out at "Source duration" while the audio still plays.

  

  FFMPEG output transcode doesn’t’ show anything odd at all. Looks normal, no errors or warnings.

  • Can I force FFMPEG to never add the "Source Duration" metadata ?
  • How do I edit the track metadata shown by mediainfo ? I tried Mp4box, and a few others, but I can’t seem to figure out how to edit track level metadata.

  Thanks.

  Log is below.

  ffmpeg version N-77455-g4707497 Copyright (c) 2000-2015 the FFmpeg developers   built with gcc 5.2.0 (GCC)   configuration: --enable-gpl
  
      --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --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-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --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. 11.100 / 55. 11.100   libavcodec     57. 20.100 / 57. 20.100   libavformat    57. 20.100 / 57. 20.100   libavdevice    57.  0.100 / 57.  0.100   libavfilter     6. 21.101 / 
  
      6. 21.101   libswscale      4.  0.100 /  4.  0.100   libswresample   2.  0.101 /  2.  0.101   libpostproc    54.  0.100 / 54.  0.100 Input #0, avi, from 'E:\MEDIA\Central.Intelligence.2016.HC.HDRip.XviD.AC3-EVO\Central.Intelligence.2016.HC.HDRip.XviD.AC3-EVO.avi': Metadata:
  
          encoder         : VirtualDubMod 1.5.10.2 (build 2542/release)   Duration: 01:51:43.27, start: 0.000000, bitrate: 1765 kb/s
  
          Stream #0:0: Video: mpeg4 (Advanced Simple Profile) (XVID / 0x44495658), yuv420p, 720x304 [SAR 1:1 DAR 45:19], 1563 kb/s, 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc
  
          Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp, 192 kb/s [libx264 @ 000000f3c80a0a80] using SAR=405/304 [libx264 @ 000000f3c80a0a80] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 000000f3c80a0a80] profile High, level 3.1 [libx264 @ 000000f3c80a0a80] 264 - core 148 r2638 7599210 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=1280 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'F:\STREAMS\Central.Intelligence.2016.HC.HDRip.XviD.AC3-EVO.avi.1280x720_1000kbps.TEMP.mp4': Metadata:
  
          encoder         : Lavf57.20.100
  
          Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 405:304 DAR 45:19], q=-1--1, 1280 kb/s, 23.98 fps, 10000k tbn, 23.98 tbc
  
          Metadata:
  
            encoder         : Lavc57.20.100 libx264
  
          Side data:
  
            unknown side data type 10 (24 bytes)
  
          Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 160 kb/s
  
          Metadata:
  
            encoder         : Lavc57.20.100 aac Stream mapping:   Stream #0:0 -> #0:0 (mpeg4 (native) -> h264 (libx264))   Stream #0:1 -> #0:1 (ac3 (native) -> aac (native)) Press [q] to stop, [?] for help [mpeg4 @ 000000f3c80ae220] Video uses a non-standard and wasteful way to store B-frames ('packed B-frames'). Consider using the mpeg4_unpack_bframes bitstream filter without encoding but stream copy to fix it. 
  
  frame=   15 fps=0.0 q=0.0 size=       0kB time=00:00:01.18 bitrate=   0.3kbits/s dup=1 drop=0 speed=2.36x   
  
  --SNIP---
  
  frame=160704 fps= 16 q=30.0 size= 1180376kB time=01:51:43.08 bitrate=1442.6kbits/s dup=1 drop=0 speed=0.671x     
  
  [libx264 @ 0000001f60c5d900] frame I:1167  Avg QP:19.08  size: 44465
  
  [libx264 @ 0000001f60c5d900] frame P:63049 Avg QP:22.26  size: 12774
  
  [libx264 @ 0000001f60c5d900] frame B:110691 Avg QP:24.81  size:  2739
  
  [libx264 @ 0000001f60c5d900] consecutive B-frames:  6.0% 22.8% 18.3% 53.0%
  
  [libx264 @ 0000001f60c5d900] mb I  I16..4: 14.0% 76.1%  9.9%
  
  [libx264 @ 0000001f60c5d900] mb P  I16..4:  2.5%  6.6%  0.4%  P16..4: 44.5% 10.8%  5.2%  0.0%  0.0%    skip:30.0%
  
  [libx264 @ 0000001f60c5d900] mb B  I16..4:  0.1%  0.3%  0.0%  B16..8: 32.1%  1.6%  0.2%  direct: 0.9%  skip:64.9%  L0:37.9% L1:58.7% BI: 3.5%
  
  [libx264 @ 0000001f60c5d900] final ratefactor: 23.25
  
  [libx264 @ 0000001f60c5d900] 8x8 transform intra:70.8% inter:85.8%
  
  [libx264 @ 0000001f60c5d900] coded y,uvDC,uvAC intra: 47.1% 56.6% 19.5% inter: 10.7% 14.2% 0.4%
  
  [libx264 @ 0000001f60c5d900] i16 v,h,dc,p: 46% 20%  8% 26%
  
  [libx264 @ 0000001f60c5d900] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 17% 25%  5%  7%  7%  7%  6%  6%
  
  [libx264 @ 0000001f60c5d900] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 20% 12%  5% 10%  9%  8%  5%  4%
  
  [libx264 @ 0000001f60c5d900] i8c dc,h,v,p: 58% 18% 18%  6%
  
  [libx264 @ 0000001f60c5d900] Weighted P-Frames: Y:1.4% UV:0.8%
  
  [libx264 @ 0000001f60c5d900] ref P L0: 62.6% 12.9% 18.9%  5.6%  0.1%
  
  [libx264 @ 0000001f60c5d900] ref B L0: 87.6% 11.0%  1.4%
  
  [libx264 @ 0000001f60c5d900] ref B L1: 94.6%  5.4%
  
  [libx264 @ 0000001f60c5d900] kb/s:1272.59
  
  [aac @ 0000001f60b52180] Qavg: 954.859

  Edit

  This has something to do with data, when I run this command (cut at 10 minutes)

  ffmpeg.exe -i CI.mp4 -ss 00:00:00 -t 00:08:00 -vcodec copy  -map_metadata 0 -acodec copy CI2.mp4

  Mediainfo shows

  Duration                    : 9mn 49s
  
  Source duration             : 2mn 40s

  When I run this one (18 minutes)

  ffmpeg.exe -i CI.mp4 -ss 00:00:00 -t 00:18:00 -vcodec copy  -map_metadata 0 -acodec copy CI2.mp4

  Mediainfo shows

  Duration                    : 17mn 49s
  
  Source duration             : 3mn 30s

  How can I edit the metadata directly