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• vcodec copy does not work when start/duration is messed up in video file

  25 janvier 2014, par anishsane

  My Video file shows below meta-data with ffprobe/ffmpeg :

  Duration: 00:44:27.52, start: 1333.760000, bitrate: 335 kb/s
  
    Stream #0.0(und): Video: h264 (Main), yuv420p, 640x480, 25 tbr, 90k tbn, 50 tbc

  Note : The file does not contain audio.

  I am trying to convert this video file to other video file, using ffmpeg/avconv.

  This works : (but encodes h.264 video to mpeg4)

  ffmpeg -i input.mp4 output.mp4

  & it generates output file of proper duration (44:27 - 1333 seconds = 22:14)

  This does not work :

  ffmpeg -i input.mp4 -vcodec copy output.mp4

  Generates file without video.

  The output contains :

  $ avconv -i input.mp4 -vcodec copy output.mp4
  
  
  
  avconv version 0.8.9-6:0.8.9-0ubuntu0.13.10.1, Copyright (c) 2000-2013 the Libav developers
  
    built on Nov  9 2013 19:09:46 with gcc 4.8.1
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4':
  
    Metadata:
  
      major_brand     : dash
  
      minor_version   : 0
  
      compatible_brands: iso6avc1mp41
  
      creation_time   : 2014-01-19 22:43:21
  
    Duration: 00:44:27.52, start: 1333.760000, bitrate: 335 kb/s
  
      Stream #0.0(und): Video: h264 (Main), yuv420p, 640x480, 25 tbr, 90k tbn, 50 tbc
  
      Metadata:
  
        creation_time   : 2014-01-19 22:43:21
  
  Output #0, mp4, to 'output.mp4':
  
    Metadata:
  
      major_brand     : dash
  
      minor_version   : 0
  
      compatible_brands: iso6avc1mp41
  
      creation_time   : 2014-01-19 22:43:21
  
      encoder         : Lavf53.21.1
  
      Stream #0.0(und): Video: ![0][0][0] / 0x0021, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
  
      Metadata:
  
        creation_time   : 2014-01-19 22:43:21
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press ctrl-c to stop encoding
  
  frame=    0 fps=  0 q=-1.0 Lsize=       0kB time=10000000000.00 bitrate=   0.0kbits/s    
  
  video:0kB audio:0kB global headers:0kB muxing overhead inf%

• 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.

  


• 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.