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• mp4fragment generates negative start time

  23 janvier 2018, par Benson Chang

  I found that when I use mp4fragment to generate fragmented mp4 file, ffmpeg gives negative start time rather than zero start time.

  My question is :

  1. Is this correct
  2. If not how should I fix it ?

  Following is the log of ffprobe for the original video.

  ffprobe version 3.4.1-1~16.04.york0 Copyright (c) 2007-2017 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.5) 20160609
  
    configuration: --prefix=/usr --extra-version='1~16.04.york0' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --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-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --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-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Waterside_Trail.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.32.100
  
    Duration: 00:05:01.12, start: 0.000000, bitrate: 13495 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 4096x2048 [SAR 1:1 DAR 2:1], 13358 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler

  The command to fragment mp4

   mp4fragment Waterside_Trail.mp4 Waterside_Trail_frag.mp4

  FFprobe result of the fragmented mp4

  ffprobe version 3.4.1-1~16.04.york0 Copyright (c) 2007-2017 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.5) 20160609
  
    configuration: --prefix=/usr --extra-version='1~16.04.york0' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --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-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --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-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Waterside_Trail_frag.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41iso5
  
   *Duration: 00:05:01.12, start: -0.023220, bitrate: 13489 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 4096x2048 [SAR 1:1 DAR 2:1], 13358 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
  
      Metadata:
  
        handler_name    : Bento4 Video Handler
  
      Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
  
      Metadata:
  
        handler_name    : Bento4 Sound Handler

  And you can see the line with asterisk, there’s a negative start time of -0.023220.

• Ffmpeg converting to mp4 from mkv[mpeg@ 0x7f9c19800000] start time for stream 0 is not set in estimate_timings_from_pts

  6 février 2018, par croakouttatune

  I recently converted a number of anime episodes from MKV to MP4 to burn and watch on my SamsungBRPlayer, however during the process I wasn’t able to convert the subtitle stream #0:2 through ffmpeg from .ssa to .srt thru their respective codecs (SSA to MOV_TEXT, can also be SUBRIP). I eventually decided to extract the SSA files and encode them as .srt... one for each episode. I converted these to .srt and plugged them back into the 8 episodes.

  for i in *.mkv;do ffmpeg -i "$i" -i *.srt -c copy -c:s mov_text -c:v h264_videotoolbox -c:a aac -b:a 128k -target ntsc-dvd -y "yfolder/${i%.mkv*}.mp4"; done

  After testing the compatibility of these files I know that this video codec will work, and the BluRay player I use also recognizes the subtitle files ; However , when looking back at the streams #0:0 which is where the subtitles are stored gives me "[mpeg @ 0x7f8621000000] start time for stream 0 is not set in estimate_timings_from_pts." This stream #0:O is now a data stream...0:1 video 0:2 being audio ...
  One of the reasons I thought that I could have received this message is from an Attachment from the original MKV files in the #0:3 stream, which because it wasn’t anything more than metadata I ignored.
  Another would probably be from the code mentioned above importing multiple .srt files into each of the new .mp4 files. I did find a solution however I’m not able to utilize the coding.

  $ for video in *.mkv 
  
  do 
  
  base=${video%.mkv} ffmpeg -i $base.mkv -vf subtitles=$base.srt  
  
  $base-out.mkv ; done

  I couldn’t seem to get it to work. My files were as follows :

  [AnimeGT] Hansom Gold - 001 [720p] [suitup].mp4

  [AnimeGT] Hansom Gold - 002 [720p] [suitup].mp4

  [AnimeGT] Hansom Gold - 003 [720p] [suitup].mp4

  [AnimeGT] Hansom Gold - 004 [720p] [suitup].mp4

  [AnimeGT] Hansom Gold - 005 [720p] [suitup].mp4

  [AnimeGT] Hansom Gold - 006 [720p] [suitup].mp4

  [AnimeGT] Hansom Gold - 007 [720p] [suitup].mp4

  [AnimeGT] Hansom Gold - 008 [720p] [suitup].mp4

  So the titles are obviously fake but whatever. What I need to know is how to use the ${I%.*}.mp4 and ${h%.*}.srt to represent both the base and the video variables in the coding I failed at above.
  While keeping the data stream from each file.
  if that’s the problem. Some help would be nice.

  [mpeg @ 0x7f9c19800000] start time for stream 0 is not set in estimate_timings_from_pts

  I need to know how to deal with this.

• How can I remove the zero padding at the start of a .mp3 file generated by ffmpeg conversion ?

  22 juin 2022, par Miguel de Sousa

  After reading this and this, I understand that the .mp3 encoder appends zeros at the start and at the end of an audio.

  


  With this approach, the encoder can pass a sliding analysis window through the initial and final audio samples, just like in the rest of the audio.

  


  This sliding window is associated with a Fourier Transform-like algorithm, but I will not get into this technical detail.

  


  My problem is : When I generate an .mp3 with ffmpeg like :

  


  MacBook-Air-de-Miguel:faixas miguel$ ffmpeg -i primeira_caf.caf primeira_agora_vai.mp3
ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 11.0.3 (clang-1103.0.32.62)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_1 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[caf @ 0x7fe205009000] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, caf, from 'primeira_caf.caf':
  Duration: 00:00:08.73, start: 0.000000, bitrate: 1414 kb/s
    Stream #0:0: Audio: pcm_s16le (lpcm / 0x6D63706C), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'primeira_agora_vai.mp3':
  Metadata:
    TSSE            : Lavf58.45.100
    Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
    Metadata:
      encoder         : Lavc58.91.100 libmp3lame
size=     137kB time=00:00:08.75 bitrate= 128.6kbits/s speed=  61x    
video:0kB audio:137kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.180156%


  


  my generated .mp3 has +- 50ms of latency. I'm not 100% sure that all this latency is caused by the zero padding that I mentioned.

  


  Converting .caf to other audio formats like .wav and .ogg do not give me this problem.

  


  Considering .mp3 as a constraint, is there any simple way to generate a .mp3 with zero latency ? Maybe a ffmpeg argument that cuts off the zero padding at the start ?

  


  If not, is it possible to do it manually ? How can I know or calculate how many samples should be cut off ?

  


  I have also tried sox. It adds 25ms of latency instead of 50ms.