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• ffmpeg : video codec ansi not compatible with flv

  11 septembre 2018, par diatomym

  lets say i have 10 video files, encoded with the following command in ffmpeg

  ffmpeg -i input.mp4 -c:v libx264 -preset medium -maxrate 6000k -bufsize 6000k -vf "scale=1280:-1,format=yuv420p" -g 50 -c:a aac -b:a 128k -ac 2 -ar 44100 file.flv

  now that all files match in codec, what i want to do is stream all those files via RTMP. for that, I’ll need to create a concat list. i also want the stream to infinitely repeat those 10 files. to do all these things, I use this command :

  ffmpeg -threads 2 -re -fflags +genpts -stream_loop -1 -i mylist.txt -c copy -f flv rtmp://link.to/RTMP

  when doing that, I get the following error output :

   ffmpeg -threads 2 -re -fflags +genpts -stream_loop -1 -i mylist.txt -c copy -f flv rtmp://link.to/RTMP
  
  ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Input #0, tty, from 'mylist.txt':
  
    Duration: 00:00:00.24, bitrate: 40 kb/s
  
      Stream #0:0: Video: ansi, pal8, 640x400, 25 fps, 25 tbr, 25 tbn, 25 tbc
  
  [flv @ 0x560ec7662920] ***Video codec ansi not compatible with flv
  
  Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented***
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)

  ffmpeg is giving me the error

  Video codec ansi not compatible with flv
Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented

  what am i doing wrong ? thanks for the help.

• Invalid data found in the last hls segment after transcoding

  28 février 2024, par GalaDOS

  When I use ffmpeg to transcode some pure audio flv files to hls segments, the last .ts segment contains some decoding errors. It seems that it happens when the last segment's duration is short enough (about 1.5 second).

  



  Here is my transcode command :

  



  ffmpeg -i 111.flv -c:a aac -b:a 128k -f hls -hls_time 10.00 -hls_list_size 9999 tmp.m3u8


  



  And output :

  



  ffmpeg version n4.0.1-5-gb5106c5 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
  configuration: --prefix=/home/zx/ffmpeg_build --extra-cflags=-I/home/zx/ffmpeg_build/include --extra-ldflags='-L/home/zx/ffmpeg_build/lib -ldl' --bindir=/home/zx/bin --pkg-config-flags=--static --enable-gpl --enable-nonfree --enable-version3 --enable-libfdk_aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libspeex --enable-libvpx --enable-libx264 --disable-shared --enable-static --disable-debug
  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
  libswscale      5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc    55.  1.100 / 55.  1.100
Input #0, flv, from '111.flv':
  Metadata:
    major_brand     : M4A 
    minor_version   : 512
    compatible_brands: isomiso2
    date            : 2018-05-31 01:31
    encoder         : Lavf57.25.100
  Duration: 00:12:11.40, start: 0.000000, bitrate: 135 kb/s
    Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
[hls @ 0x25d4280] Opening 'tmp0.ts' for writing
[mpegts @ 0x26feb80] frame size not set
Output #0, hls, to 'tmp.m3u8':
  Metadata:
    major_brand     : M4A 
    minor_version   : 512
    compatible_brands: isomiso2
    date            : 2018-05-31 01:31
    encoder         : Lavf58.12.100
    Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.18.100 aac
[hls @ 0x25d4280] Opening 'tmp1.ts' for writing
[hls @ 0x25d4280] Opening 'tmp.m3u8.tmp' for writing
[hls @ 0x25d4280] Opening 'tmp2.ts' for writing
[hls @ 0x25d4280] Opening 'tmp.m3u8.tmp' for writing
[hls @ 0x25d4280] Opening 'tmp3.ts' for writing
[hls @ 0x25d4280] Opening 'tmp.m3u8.tmp' for writing
[hls @ 0x25d4280] Opening 'tmp4.ts' for writing
......
[hls @ 0x25d4280] Opening 'tmp71.ts' for writing
[hls @ 0x25d4280] Opening 'tmp.m3u8.tmp' for writing
[hls @ 0x25d4280] Opening 'tmp72.ts' for writingx    
[hls @ 0x25d4280] Opening 'tmp.m3u8.tmp' for writing
[hls @ 0x25d4280] Opening 'tmp73.ts' for writing
[hls @ 0x25d4280] Opening 'tmp.m3u8.tmp' for writing
[hls @ 0x25d4280] Opening 'tmp.m3u8.tmp' for writing
size=N/A time=00:12:11.39 bitrate=N/A speed=84.7x    
video:0kB audio:11421kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[aac @ 0x25a0580] Qavg: 567.976


  



  After transcoding, the duration of the last segment in m3u8 file is 1.39 second :

  



  ....
#EXTINF:9.984000,
tmp71.ts
#EXTINF:10.005333,
tmp72.ts
#EXTINF:1.386667,
tmp73.ts
#EXT-X-ENDLIST


  



  But when I use ffprobe to parse the last segment, I got an error :

  



  ffprobe -i tmp73.ts 
ffprobe version n4.0.1-5-gb5106c5 Copyright (c) 2007-2018 the FFmpeg developers
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
  configuration: --prefix=/home/zx/ffmpeg_build --extra-cflags=-I/home/zx/ffmpeg_build/include --extra-ldflags='-L/home/zx/ffmpeg_build/lib -ldl' --bindir=/home/zx/bin --pkg-config-flags=--static --enable-gpl --enable-nonfree --enable-version3 --enable-libfdk_aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libspeex --enable-libvpx --enable-libx264 --disable-shared --enable-static --disable-debug
  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
  libswscale      5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc    55.  1.100 / 55.  1.100
tmp73.ts: Invalid data found when processing input


  



  The input flv :

  



  http://45.76.56.248/hls_transcode_fail.flv
md5sum: 4627bbb27bcb22c143ad4dfa47359650


  



  Since my statistic module relies on the output of ffprobe, this exception will cause the entire transcoding task to fail. My temporary solution is to skip the last segment which is too short in the statistics phase. But I need a more reasonable and robust way to fix or bypass this issue. Any suggestion will be appreciated.

  


• ffmpeg : mix/merge multiple mp3 files, some do not mix

  28 août 2018, par C. Ovidiu

  I am trying to merge multiple mp3 files on top of each other on a CentOS 7 server.

  I am trying with ffmpeg but I have mixed results. When mixing 4 files, the last one for example does not mix with the others and is not audible in the final output.

  If I mix this file with another one or two(so max 3 files merged), it works.

  Is there a limit when merging ? For reference, each file is about 10mb is size and 5:00 minutes long.

  This is the command I am using

  ffmpeg -i /var/www/vhosts/site/httpdocs/uploads/tracks/1.mp3 -i /var/www/vhosts/site/httpdocs/uploads/tracks/2.mp3 -i /var/www/vhosts/site/httpdocs/uploads/tracks/3.mp3 -i /var/www/vhosts/site/httpdocs/uploads/tracks/4.mp3 -filter_complex amerge -ac 2 -c:a libmp3lame -q:a 4 /var/www/vhosts/site/httpdocs/uploads/mix.mp3

  The output after merging is this :

  ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0x1c8ba60] Skipping 0 bytes of junk at 1044.
  
  Input #0, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/1.mp3':
  
    Duration: 00:05:44.08, start: 0.025057, bitrate: 320 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
      Metadata:
  
        encoder         : LAME3.98r
  
      Side data:
  
        replaygain: track gain - 6.000000, track peak - unknown, album gain - unknown, album peak - unknown,
  
  [mp3 @ 0x1c8eac0] Skipping 0 bytes of junk at 2446.
  
  [mp3 @ 0x1c8eac0] Estimating duration from bitrate, this may be inaccurate
  
  Input #1, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/2.mp3':
  
    Metadata:
  
      genre           : Other
  
    Duration: 00:05:44.19, start: 0.000000, bitrate: 320 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
  [mp3 @ 0x1c9d640] Skipping 0 bytes of junk at 1044.
  
  Input #2, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/3.mp3':
  
    Duration: 00:05:44.08, start: 0.025057, bitrate: 320 kb/s
  
      Stream #2:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
      Metadata:
  
        encoder         : LAME3.98r
  
      Side data:
  
        replaygain: track gain - 3.400000, track peak - unknown, album gain - unknown, album peak - unknown,
  
  [mp3 @ 0x1cc2b80] Skipping 0 bytes of junk at 1044.
  
  Input #3, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/4.mp3':
  
    Duration: 00:05:44.08, start: 0.025057, bitrate: 320 kb/s
  
      Stream #3:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
      Metadata:
  
        encoder         : LAME3.98r
  
      Side data:
  
        replaygain: track gain - 12.100000, track peak - unknown, album gain - unknown, album peak - unknown,
  
  [Parsed_amerge_0 @ 0x1cc34e0] No channel layout for input 1
  
  [Parsed_amerge_0 @ 0x1cc34e0] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
  
  Output #0, mp3, to '/var/www/vhosts/site/httpdocs/uploads/mix.mp3':
  
    Metadata:
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p (default)
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 (mp3) -> amerge:in0
  
    Stream #1:0 (mp3) -> amerge:in1
  
    amerge -> Stream #0:0 (libmp3lame)
  
  Press [q] to stop, [?] for help
  
  size=    2360kB time=00:05:44.03 bitrate=  56.2kbits/s
  
  video:0kB audio:2360kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.010468%

  Is there a way to solve this, or at least to know what the issue is ?

  Also, some people recommended sox, but I can’t figure how to install it on CentOS.

  Any other alternatives will also help.

  Thank you