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• ffmpeg demux into audio and video resets PTS

  30 juillet 2018, par Mukund Manikarnike

  Demuxing

  I am demuxing TS segments into audio and video as follows.

  ffmpeg -y -i input.ts -vcodec copy -an output_video.ts
  
  ffmpeg -y -i input.ts -acodec copy -vn output_audio.aac

  Inspecting Input

  The start_pts and start_time on input.ts are as shown below. I was able to inspect these values using ffprobe -show_streams -print_format json input.ts

  "start_pts": 8306558438,
  
  "start_time": "92295.093756",

  Inspecting output video

  The output .ts has some default start_pts and start_time values as shown below. These were also obtained using the same ffprobe command as indicated above.

  "start_pts": 126000,
  
  "start_time": "1.400000",

  Inspecting output audio

  The same ffprobe command on output_audio.aac shows that the output aac has invalid codec_tag and codec_tag_string as shown below. The start_pts and start_time are not present in the output_audio.aac.

  "codec_tag_string": "[0][0][0][0]", (should have been [15][0][0][0])
  
  "codec_tag": "0x0000", (should have been 0xf000)

  Questions

  1. Wondering if this difference in the start_pts, start_time, codec_tag is expected ?
  2. If it is expected, what can I do to ensure that the all of these parameters get retained on the output ?
  3. If it is not expected, is there some more information I can share to track this down ?

  Note

  There were other outputs that I found inconsistent in the ffprobe command for the output_audio.aac like duration etc.. I shared what I thought are most valuable at this point. If required I can share complete outputs from all of the above executions.

  [EDIT 07/30/2018 - 08:00 MST]
  logs for ffmpeg -y -i input.ts -vcodec copy -an output_video.ts -acodec copy -vn output_audio.aacare as shown below.

  ffmpeg version 4.0.2 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with Apple LLVM version 9.0.0 (clang-900.0.39.2)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-ffplay --enable-frei0r --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-librtmp --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma --enable-libopenjpeg --disable-decoder=jpeg2000 --extra-cflags=-I/usr/local/Cellar/openjpeg/2.3.0/include/openjpeg-2.3 --enable-nonfree
  
    libavutil      56. 14.100 / 56. 14.100
  
    libavcodec     58. 18.100 / 58. 18.100
  
    libavformat    58. 12.100 / 58. 12.100
  
    libavdevice    58.  3.100 / 58.  3.100
  
    libavfilter     7. 16.100 /  7. 16.100
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  1.100 /  5.  1.100
  
    libswresample   3.  1.100 /  3.  1.100
  
    libpostproc    55.  1.100 / 55.  1.100
  
  [mpegts @ 0x7f88ed803000] start time for stream 0 is not set in estimate_timings_from_pts
  
  Input #0, mpegts, from 'i7h9456s_media_46185.ts':
  
    Duration: 00:00:06.05, start: 86216.852667, bitrate: 2898 kb/s
  
    Program 1 
  
      Stream #0:0[0x102]: Data: timed_id3 (ID3  / 0x20334449)
  
      Stream #0:1[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p(tv, smpte170m, progressive), 640x360 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
  
      Stream #0:2[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 190 kb/s
  
  Output #0, mpegts, to '../output_video.ts':
  
    Metadata:
  
      encoder         : Lavf58.12.100
  
      Stream #0:0: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p(tv, smpte170m, progressive), 640x360 [SAR 1:1 DAR 16:9], q=2-31, 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
  
  Output #1, adts, to '../output_audio.aac':
  
    Metadata:
  
      encoder         : Lavf58.12.100
  
      Stream #1:0: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 190 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
    Stream #0:2 -> #1:0 (copy)
  
  Press [q] to stop, [?] for help
  
  frame=  180 fps=0.0 q=-1.0 Lsize=    2088kB time=00:00:06.03 bitrate=2833.8kbits/s speed= 904x    
  
  video:1918kB audio:142kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.349750%

• ffmpeg with "-pattern_type glob" and variable in bash script

  20 juin 2019, par KlausPeter

  I’m trying to let ffmpeg make a video of all pictures in a directory using the -pattern_type glob switch and "/foo/bar/*.jpg". This works well, if I execute the command manually für just one directory. For example :

  ffmpeg -framerate 35 -pattern_type glob -i '/root/webcam_test/2018-07-21/*.jpg' -vf scale=1280:-1 -c -c:v libx264 -pix_fmt yuv420p /root/clips/out01_cut.mp4

  However, if I do it in a bash script and set the path via a variable, according to ffmpegs output, the variable gets substituted correctly, but ffmpeg states that

  ’/root/webcam_test/2018-07-21/*.jpg’ : No such file or directory

  The part of the script looks like this :

  for D in `find /root/webcam_test/ -type d`
  
  do
  
      [...]
  
      cmd="ffmpeg -framerate 35 -pattern_type glob -i '$D/*.jpg' -vf scale=1280:-1 -c -c:v libx264 -pix_fm                                 t yuv420p /root/clips/$d_cut.mp4"
  
      echo $cmd
  
  [...]
  
  done

  Does anyone know how to make ffmpeg do its wildcard interpretation even if the path is constructed by a script and not just try to plainly use the given path ?
  Best regards and thanks in advance

• Duration of short ogg files (Telegram Voice messages) not correct when loaded into Python

  4 août 2018, par Kromme

  I’m trying to read voice messages, sent by Telegram, using Python but for short voice clips (< 10 seconds), it doesn’t work. It shortens the duration for some reason. It looks like it has something to do with OGG codec, but I’m not really sure.

  See here’s my code, the voice clip is about six seconds, however pydub reads my 6 second voiceclip as 0.06 seconds.

  import telegram
  
  from pydub import AudioSegment
  
  
  
  AudioSegment.ffmpeg = "./dependencies/ffmpeg-20180802-c9118d4-win64-static/bin/ffmpeg"
  
  AudioSegment.converter = "./dependencies/ffmpeg-20180802-c9118d4-win64-static/bin/ffmpeg"
  
  
  
  
  
  bot = telegram.Bot(token=token)
  
  f = bot.get_file(file_id)
  
  f.download('output/voiceclips/{}.ogg'.format(file_id))
  
  
  
  myaudio = AudioSegment.from_ogg("output/voiceclips/{}.ogg".format(file_id))
  
  print('ID: {}, which is {} seconds'.format(file_id, myaudio.duration_seconds)) 
  
  
  
  >>> ID: ______, which is 0.06 seconds

  When I open the file in VLC-player, it also states that is has 0 seconds. When I try to convert it to WAV-files using FFmpeg it reads the ogg file as 6 seconds, but writes it as 0.05-second WAV file.

  ffmpeg -i infile.ogg outfile.wav
  
  ffmpeg version N-91549-gc9118d4d64 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.1 (GCC) 20180722
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 18.102 / 56. 18.102
  
    libavcodec     58. 22.100 / 58. 22.100
  
    libavformat    58. 17.101 / 58. 17.101
  
    libavdevice    58.  4.101 / 58.  4.101
  
    libavfilter     7. 26.100 /  7. 26.100
  
    libswscale      5.  2.100 /  5.  2.100
  
    libswresample   3.  2.100 /  3.  2.100
  
    libpostproc    55.  2.100 / 55.  2.100
  
  [ogg @ 0000020dd375ad40] 727 bytes of comment header remain
  
  Input #0, ogg, from 'infile.ogg':
  
    Duration: 00:00:06.03, start: 0.000000, bitrate: 20 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, wav, to 'outfile.wav':
  
    Metadata:
  
      ISFT            : Lavf58.17.101
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s
  
      Metadata:
  
        encoder         : Lavc58.22.100 pcm_s16le
  
  size=       6kB time=00:00:00.05 bitrate= 873.0kbits/s speed=4.12x
  
  video:0kB audio:6kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.354167%

  For larger files it does the work !