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  • ffmpeg default audio codec instead of specifying it with acodec option

    28 août 2020, par Kazooie

    In raspberry pi I've following i2s microphone breakout board and use it like the guide suggested. When I try record audio from it using ffmpeg to the file with ffmpeg -f alsa -i dmic_sv out.wav command. I'll receive following error

    


    [alsa @ 0x22e21c0] cannot set sample format 0x10000 2 (Invalid argument)
dmic_sv: Input/output error


    


    When I specify the used codec explicitly with -acodec it works fine :

    


    ffmpeg -f alsa -acodec pcm_s32le -i dmic_sv out.wav


    


    And from the output ffmpeg will reencode to pcm_s16le

    


    Input #0, alsa, from 'dmic_sv':
  Duration: N/A, start: 1597597938.887969, bitrate: 3072 kb/s
    Stream #0:0: Audio: pcm_s32le, 48000 Hz, stereo, s32, 3072 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s32le (native) -> pcm_s16le (native))


    


    How I can tell ffmpeg to use signed 32-bit little endian by default without specifying it explicitly ? And where ffmpeg gets this default and can I configure it somehow ?

    


  • Unable to find a suitable output format for 'ffmpeg' I/flutter (29205) : ffmpeg : Invalid argument

    24 juillet 2020, par Abhijit Rajmane

    I'm trying to mix two audio files using flutter_ffmpeg : 0.2.10 but I get

    


    [NULL @ 0xd77d4400] Unable to find a suitable output format for 'ffmpeg'
I/flutter (29205): ffmpeg: Invalid argument

code : 
String file1= _localPath+"/file1.m4a";
String file2= _localPath+"/file2.mp3";
String output= _localPath+"/abhijit.wav";

String finalPath= "ffmpeg -i "+ file1 +" -i "+file2+" -filter_complex amix=inputs=2:duration=first:dropout_transition=0 "+output;

console ffmpeg's message...
I/flutter (29205):   libavutil      56. 38.100 / 56. 38.100
I/flutter (29205):   libavcodec     58. 65.102 / 58. 65.102
I/flutter (29205):   libavformat    58. 35.101 / 58. 35.101
I/flutter (29205):   libavdevice    58.  9.103 / 58.  9.103
I/flutter (29205):   libavfilter     7. 70.101 /  7. 70.101
I/flutter (29205):   libswscale      5.  6.100 /  5.  6.100
I/flutter (29205):   libswresample   3.  6.100 /  3.  6.100
I/flutter (29205): Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/Android/data/com.arthenica.flutter.ffmpeg.FlutterFFmpegExample/files/Download/file1.m4a':
I/flutter (29205):   Metadata:
I/flutter (29205):     major_brand     : mp42
I/flutter (29205):     minor_version   : 1
I/flutter (29205):     compatible_brands: isommp41mp42
I/flutter (29205):     creation_time   : 2020-07-09T15:42:47.000000Z
I/flutter (29205):   Duration: 00:12:12.26, start: 0.000000, bitrate: 61 kb/s
I/flutter (29205):     Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 60 kb/s (default)
I/flutter (29205):     Metadata:
I/flutter (29205):       creation_time   : 2020-07-09T15:42:47.000000Z
I/flutter (29205):       handler_name    : Core Media Audio
I/flutter (29205): Input #1, mp3, from '/storage/emulated/0/Android/data/com.arthenica.flutter.ffmpeg.FlutterFFmpegExample/files/Download/file2.mp3':
I/flutter (29205):   Metadata:
I/flutter (29205):     artist          : SoundHelix
I/flutter (29205):     genre           : Techno
I/flutter (29205):     encoder         : Lavf58.20.100
I/flutter (29205):   Duration: 00:00:23.12, start: 0.011995, bitrate: 192 kb/s
I/flutter (29205):     Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
I/flutter (29205):     Metadata:
I/flutter (29205):       encoder         : Lavf
I/flutter (29205): [NULL @ 0xd77d4400] Unable to find a suitable output format for 'ffmpeg'
I/flutter (29205): ffmpeg: Invalid argument
I/flutter (29205): "
I/flutter (29205): New FFmpeg pipe: /data/user/0/com.arthenica.flutter.ffmpeg.FlutterFFmpegExample/cache/mf_pipe_7
I/flutter (29205): FFmpeg process exited with rc 1


    


    Please check command of FFmpeg and suggest me if any correction occur, Unable to find a suitable output format for 'ffmpeg' I/flutter (29205) : ffmpeg : Invalid argument
I have no idea about this error please help me

    


  • 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 :

    



    VLC shows as WebVTT

    



    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 ...

    


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