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• Set correct start time of ts-file using ffmpeg

  9 octobre 2020, par Daniel

  I am splitting up a video into multiple 10 second ts-parts (mpeg-ts format) using ffmpeg on windows.

  



  To create the 2nd part (that starts at 10 seconds into the video and ends at 20 seconds into the video) :

  



  ffmpeg -i sample.avi -ss 00:00:10 -to 00:00:20 -vcodec libx264 -acodec aac -vf scale=426:-1 out1.ts


  



  But when i check the file using ffprobe it says :

  



  Duration: 00:00:10.02, start: 1.458667, bitrate: 359 kb/s


  



  So the duration is ok but the start time is incorrect. Is it anyway i can use ffmpeg to correct it to 00:00:20 ?
The best solution would of course to be able to set the correct start time in my first command where i take out the 10 second part but i would also be ok with running a 2nd command to fix the time.

  



  Is this possible ? Cant find any documentation and all examples i found are not for my exact problem and don't seem to work then i play around with them.

  



  Full output from ffprobe :

  



  ffprobe.exe out1.ts
ffprobe version git-2020-02-06-343ccfc Copyright (c) 2007-2020 the FFmpeg developers
  built with gcc 9.2.1 (GCC) 20200122
  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. 39.100 / 56. 39.100
  libavcodec     58. 68.100 / 58. 68.100
  libavformat    58. 38.100 / 58. 38.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 74.100 /  7. 74.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, mpegts, from 'out1.ts':
  Duration: 00:00:10.02, start: 1.458667, bitrate: 359 kb/s
  Program 1
    Metadata:
      service_name    : Service01
      service_provider: FFmpeg
    Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 426x260 [SAR 780:781 DAR 18:11], 25 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 131 kb/s


  


• Compilied Ffmpeg not accepting -c:v and -c:a

  2 février 2020, par King Horse

  I complied FFMPEG with libsrt, with the online compile guide. https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu & how to compile ffmpeg with enabling libsrt

  It seems to compile correctly.

  ffmpeg version N-96575-g843c24a Copyright (c) 2000-2020 the FFmpeg developers
  
  built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
  
  configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ubuntu/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/ubuntu/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libsrt --enable-nonfree
  
  libavutil      56. 38.100 / 56. 38.100
  
  libavcodec     58. 67.100 / 58. 67.100
  
  libavformat    58. 37.100 / 58. 37.100
  
  libavdevice    58.  9.103 / 58.  9.103
  
  libavfilter     7. 72.100 /  7. 72.100
  
  libswscale      5.  6.100 /  5.  6.100
  
  libswresample   3.  6.100 /  3.  6.100
  
  libpostproc    55.  6.100 / 55.  6.100

  But when running this command to convert a incoming srt stream to HLS, it doesn’t know the -c:a command. When switching the order, it runs that it doesn’t know about the -c:v command.

  ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316&mode=listener -c:a copy -c:v copy -strict -f hls -hls_time 4 -hls_playlist_type event stream.m3u8

  ~$ ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316&mode=listener -c:a copy -c:v copy -strict -f hls -hls_time 4 -hls_playlist_type event stream.m3u8
  
  [2] 9930
  
  ffmpeg version N-96575-g843c24a Copyright (c) 2000-2020 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
  
    configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ubuntu/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/ubuntu/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libsrt --enable-nonfree
  
    libavutil      56. 38.100 / 56. 38.100
  
    libavcodec     58. 67.100 / 58. 67.100
  
    libavformat    58. 37.100 / 58. 37.100
  
    libavdevice    58.  9.103 / 58.  9.103
  
    libavfilter     7. 72.100 /  7. 72.100
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  -c:a: command not found
  
  
  
  [2]+  Stopped                 ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316

  I have searched the issue, but I could not find anything similar.
  Does someone what I have missed in the setup ?

  Everything is manual complied through the guide, this was the final command I run to compile FFMPEG :

  cd ~/ffmpeg_sources && \
  
  wget -O ffmpeg-snapshot.tar.bz2 https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 && \
  
  tar xjvf ffmpeg-snapshot.tar.bz2 && \
  
  cd ffmpeg && \
  
  PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
  
    --prefix="$HOME/ffmpeg_build" \
  
    --pkg-config-flags="--static" \
  
    --extra-cflags="-I$HOME/ffmpeg_build/include" \
  
    --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  
    --extra-libs="-lpthread -lm" \
  
    --bindir="$HOME/bin" \
  
    --enable-gpl \
  
    --enable-libaom \
  
    --enable-libass \
  
    --enable-libfdk-aac \
  
    --enable-libfreetype \
  
    --enable-libmp3lame \
  
    --enable-libopus \
  
    --enable-libvorbis \
  
    --enable-libvpx \
  
    --enable-libx264 \
  
    --enable-libx265 \
  
    --enable-libsrt \
  
    --enable-nonfree && \
  
  PATH="$HOME/bin:$PATH" make && \
  
  make install && \
  
  hash -r

• Create HLS streamable audio file from mp3

  15 août 2023, par isADon

  I am using following command to create a hls aac audio file for web streaming

  



  ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8


  



  This command works only with some audio files. With many mp3 files I receive following output :

  



  C:\ffmpeg>ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8
ffmpeg version git-2020-01-31-62d92a8 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.2.1 (GCC) 20200122
  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. 38.100 / 56. 38.100
  libavcodec     58. 67.100 / 58. 67.100
  libavformat    58. 37.100 / 58. 37.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 72.100 /  7. 72.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[mp3 @ 0000027d800babc0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'song.mp3':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
  Duration: 00:03:21.46, start: 0.000000, bitrate: 325 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
    Stream #0:1: Video: mjpeg (Baseline), yuvj444p(pc, bt470bg/unknown/unknown), 400x400 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
Stream mapping:
  Stream #0:1 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Stream #0:0 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
[hls @ 0000027d80100c40] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
[libx264 @ 0000027d800c1280] using SAR=1/1
[libx264 @ 0000027d800c1280] MB rate (56250000) > level limit (16711680)
[libx264 @ 0000027d800c1280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000027d800c1280] profile High 4:4:4 Predictive, level 6.2, 4:4:4, 8-bit
[libx264 @ 0000027d800c1280] 264 - core 159 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, hls, to 'playlist.m3u8':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
    encoder         : Lavf58.37.100
    Stream #0:0: Video: h264 (libx264), yuvj444p(pc, progressive), 400x400 [SAR 72:72 DAR 1:1], q=-1--1, 90k fps, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
      encoder         : Lavc58.67.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
    Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.67.100 aac
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6 speed=68.6x
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -5 -5
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -2 -2
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -3 -3
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -4 -4
[hls @ 0000027d80100c40] Opening 'file0.m4a' for writingate=N/A speed=64.1x
[hls @ 0000027d80100c40] Opening 'playlist.m3u8.tmp' for writing
frame=    1 fps=0.3 q=33.0 Lsize=N/A time=00:03:21.45 bitrate=N/A speed=63.7x
video:7kB audio:3209kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[libx264 @ 0000027d800c1280] frame I:1     Avg QP:34.64  size:  6567
[libx264 @ 0000027d800c1280] mb I  I16..4: 19.5% 53.0% 27.5%
[libx264 @ 0000027d800c1280] 8x8 transform intra:53.0%
[libx264 @ 0000027d800c1280] coded y,u,v intra: 46.8% 26.1% 15.3%
[libx264 @ 0000027d800c1280] i16 v,h,dc,p: 38% 39%  9% 14%
[libx264 @ 0000027d800c1280] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 26%  8%  5%  6%  5%  7%  7%
[libx264 @ 0000027d800c1280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 42% 16% 14%  7%  4%  5%  3%  4%  4%
[libx264 @ 0000027d800c1280] kb/s:4728240.00
[aac @ 0000027d800bcc40] Qavg: 2138.508


  



  Notice the "mp3float overread" message.

  



  It results in a single file0.m4a file without splitting it up after every 7 seconds as specified.
This is an example audio file I am trying to convert to a aac hls stream that results the mentioned problem : https://incompetech.com/music/royalty-free/index.html?isrc=USUAN1100474

  



  How can I convert an audio file to a web friendly hls stream with ffmpeg ?