[FFmpeg-user] trimmed audio has wrong duration

[Felix Muster]

Hello

I'm trying to trim 1.001 seconds from the beginning of a FLAC file.
But the output has the wrong duration (same as the input) and so it's
corrupt.
When I try to open it with Adobe Audition I get an error (cannot read file).

General
Complete name                            : C:\_audio.flac
Format                                   : FLAC
Format/Info                              : Free Lossless Audio Codec
File size                                : 759 MiB
Duration                                 : 1 h 31 min
Overall bit rate mode                    : Variable
Overall bit rate                         : 1 160 kb/s
Track name                               : MILLENIUM_D1_VERBLENDUNG.Title0
Writing application                      : Lavf58.20.100

Audio
Format                                   : FLAC
Format/Info                              : Free Lossless Audio Codec
Duration                                 : 1 h 31 min
Bit rate mode                            : Variable
Bit rate                                 : 1 160 kb/s
Channel(s)                               : 6 channels
Channel layout                           : L R C LFE Ls Rs
Sampling rate                            : 48.0 kHz
Bit depth                                : 16 bits
Compression mode                         : Lossless
Stream size                              : 759 MiB (100%)
Writing library                          : Lavf58.20.100

_ffprobe.exe -v 0 -sexagesimal -show_entries format^=duration -of
compact^=p^=0^:nk^=1 _audio.flac
1:31:29.848938


5489.848938 - 1.001 = 5488.847938


_ffmpeg.exe -ss 1.001 -i _audio.flac -t 5488.847938 -codec copy
_audio_1.flac
ffmpeg version N-95216-ge6625ca41f Copyright (c) 2000-2019 the FFmpeg
developers
  built with gcc 9.2.1 (GCC) 20190918
  configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libdav1d --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-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt --enable-amf
  libavutil      56. 35.100 / 56. 35.100
  libavcodec     58. 59.101 / 58. 59.101
  libavformat    58. 33.100 / 58. 33.100
  libavdevice    58.  9.100 / 58.  9.100
  libavfilter     7. 61.100 /  7. 61.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, flac, from '_audio.flac':
  Metadata:
    TITLE           : MILLENIUM_D1_VERBLENDUNG.Title0
    encoder         : Lavf58.20.100
  Duration: 01:31:29.85, start: 0.000000, bitrate: 1159 kb/s
    Stream #0:0: Audio: flac, 48000 Hz, 5.1(side), s16
Output #0, flac, to '_audio_1.flac':
  Metadata:
    TITLE           : MILLENIUM_D1_VERBLENDUNG.Title0
    encoder         : Lavf58.33.100
    Stream #0:0: Audio: flac, 48000 Hz, 5.1(side), s16
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
video:0kB audio:777322kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.001046%


I also tried -t as an input option.
Same problem.

Best
Felix