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• ffmpeg doesn't detect the subtitle format (webvtt) and refuses to copy it

  29 novembre 2023, par Thomas

  I am trying to re-encode the audio on some video files.
I want to copy the video and subtitles without touching them, however, ffmpeg doesn't detect the format of the subtitle (which is webvtt) and refuses to do anything.

  



  fmpeg -copy_unknown -analyzeduration 2000M -probesize 2000M -i input.mkv -map_metadata 0 -map 0 -c copy -c:a:0 aac -b:a 384k output.mkv

  



  [matroska,webm @ 0x5597558e4040] Unknown/unsupported AVCodecID S_TEXT/WEBVTT.
[matroska,webm @ 0x5597558e4040] Could not find codec parameters for stream 2 (Subtitle: none): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Guessed Channel Layout for Input Stream #0.1 : 5.1
Input #0, matroska,webm, from 'input.mkv':
  Metadata:
    encoder         : libebml v1.3.9 + libmatroska v1.5.2
    creation_time   : 2020-03-05T09:03:02.000000Z
  Duration: 00:56:33.26, start: 0.000000, bitrate: 3597 kb/s
    Stream #0:0: Video: h264 (High), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
    Metadata:
      BPS-eng         : 3340067
      DURATION-eng    : 00:56:33.140000000
      NUMBER_OF_FRAMES-eng: 81354
      NUMBER_OF_BYTES-eng: 1416664616
      _STATISTICS_WRITING_APP-eng: mkvmerge v38.0.0 ('The Silent Type') 64-bit
      _STATISTICS_WRITING_DATE_UTC-eng: 2020-03-05 09:03:02
      _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
    Stream #0:1: Audio: eac3, 48000 Hz, 5.1, fltp (default)
    Metadata:
      BPS-eng         : 256000
      DURATION-eng    : 00:56:33.248000000
      NUMBER_OF_FRAMES-eng: 106039
      NUMBER_OF_BYTES-eng: 108583936
      _STATISTICS_WRITING_APP-eng: mkvmerge v38.0.0 ('The Silent Type') 64-bit
      _STATISTICS_WRITING_DATE_UTC-eng: 2020-03-05 09:03:02
      _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
    Stream #0:2(eng): Subtitle: none (default)
    Metadata:
      BPS-eng         : 38
      DURATION-eng    : 00:54:47.388000000
      NUMBER_OF_FRAMES-eng: 514
      NUMBER_OF_BYTES-eng: 15766
      _STATISTICS_WRITING_APP-eng: mkvmerge v38.0.0 ('The Silent Type') 64-bit
      _STATISTICS_WRITING_DATE_UTC-eng: 2020-03-05 09:03:02
      _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES

Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (eac3 (native) -> aac (native))
  Stream #0:2 -> #0:2 (copy)

[aac @ 0x5597b06ae000] Using a PCE to encode channel layout "5.1(side)"
[matroska @ 0x5597b06ab540] Subtitle codec 0 is not supported.
Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented
Error initializing output stream 0:1 -- 
[aac @ 0x5597b06ae000] Qavg: -nan
Conversion failed!


  



  VLC shows it as WebVTT and displays the subtitles correctly :

  



  

  



  MediaInfo also detects properly as webvtt :

  



  Text
ID                                       : 3
Format                                   : S_TEXT/WEBVTT
Codec ID                                 : S_TEXT/WEBVTT
Duration                                 : 54 min 47 s
Bit rate                                 : 38 b/s
Count of elements                        : 514
Stream size                              : 15.4 KiB (0%)
Language                                 : English
Default                                  : Yes
Forced                                   : No


  



  My ffmpeg was built with webvtt :

  



  $ ffmpeg -codecs | grep VTT
ffmpeg version n4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.3.0 (Arch Linux 9.3.0-1)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
 DES... webvtt               WebVTT subtitle


  



  How can I tell ffmpeg "this is webvtt, just copy it please !" ?

  



  EDIT : RESOLVED (thanks @Gyan)

  



  To force the decoder, -c:s webvtt must be set before the input :
fmpeg -analyzeduration 50M -probesize 50M -c:s webvtt -i input.mkv ...

  


• How to use ffmpeg to overlay waveforms on xstack mosaics and specify specific audio for playback

  1er mai 2022, par kellib

  I would like to make a mosaic of multiple titled streams, 1) specifying which of the audio streams to play and 2) overlay waveforms at the bottom of each of the video tiles for the audio that they belong to.

  


  I'm successfully able to create the titled mosaic of streams with the code below.

  


  However :

  


  

  1. I'm having a hard time figuring out how to specify just one of the specific audio sources. I found amix, but I don't really want to mix them, I just want to specify audio [a0], or [a1], or [a2], etc.

  


  


  and

  


  

  2. I'm having a hard time figuring out how to overlay the wave forms at the bottom of the video for each of the tiles. I struggled trying to figure out putting showwaves into the mix. Is it possible ?

  


  


  I want each tile to look like this, but since these are rtmp streams, they need to play-out the matching waveforms dynamically with each stream. https://dragonquest64.blogspot.com/2020/01/ffmpeg-audio-waveform.html

  


  If someone could point me in the right direction, that would be great. I'm getting close, but I'm pretty new to all of this, and have already spent way more time than I should have, so would love a little help.

  


  ffmpeg \
-i rtmp://my.cdn.com/srcEncoders/STREAM-1 \
-i rtmp://my.cdn.com/srcEncoders/STREAM-2 \
-i rtmp://my.cdn.com/srcEncoders/STREAM-3 \
-i rtmp://my.cdn.com/srcEncoders/STREAM-4 \
  -filter_complex " \
      [0:v] setpts=PTS-STARTPTS, scale=qvga \
    , drawtext=text=STREAM-1:fontsize=20:x=10:y=10:fontcolor=white:box=1:boxcolor=black@0.5:boxborderw=5 [a0]; \
      [1:v] setpts=PTS-STARTPTS, scale=qvga \
    , drawtext=text=STREAM-2:fontsize=20:x=10:y=10:fontcolor=white:box=1:boxcolor=black@0.5:boxborderw=5 [a1]; \
      [2:v] setpts=PTS-STARTPTS, scale=qvga \
    , drawtext=text=STREAM-3:fontsize=20:x=10:y=10:fontcolor=white:box=1:boxcolor=black@0.5:boxborderw=5 [a2]; \
      [3:v] setpts=PTS-STARTPTS, scale=qvga \ 
    , drawtext=text=STREAM-4:fontsize=20:x=10:y=10:fontcolor=white:box=1:boxcolor=black@0.5:boxborderw=5 [a3]; \
      [a0][a1][a2][a3]xstack=inputs=4:layout=0_0|0_h0|w0_0|w0_h0[out]; \
    amix=inputs=1
      " \
  -map "[out]" \
 -c:v libx264 -b:v 1000k -g 30 -keyint_min 120 -profile:v baseline -preset veryfast -f mpegts "udp://127.0.0.1:1234?pkt_size=1316"


  


• How to convert from .srt to .ass with FFmpeg without decoding errors

  10 avril 2020, par symonxd

  When I try the following command with FFmpeg (v. git-2020-03-24-e5d25d1) : ffmpeg -i subs.srt subs.ass, I get a bunch of the following errors :

  



  [srt @ 0000028ea78fa780] Invalid UTF-8 in decoded subtitles text; maybe missing -sub_charenc option
Error while decoding stream #0:0: Invalid data found when processing input


  



  and :

  



  [ass @ 0000028ea78ffe40] ReadOrder gap found between 3 and 4


  



  I found that my .srt file contains the following char : ë
I tried adding the flag -sub_charenc UTF-8 as stated here with no success.

  



  How can I successfully convert .srt to .ass ?