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• 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

• FFMPEG failing in AWS Lambda

  18 février 2019, par Zaid Amir

  I am trying to create a transcoding function for short videos. The function is hosted on AWS Lambda. The problem is that AWS lambda seems to be missing something that FFMPEG requires, at least according to Amazon.

  I contacted Amazon earlier and this is their response to the issue :

  We found that the FFMPEG operations require at least libx264 and an
  acc library, both of which will have dependencies of their own. To
  troubleshoot the issue it will involve diving deeper into the full
  dependency chain. We can see that it works in the Amazon Linux
  environment however, the environment is similar but not identical to
  the lambda environment. There can be some dependencies that exist in
  Amazon Linux but not in lambda environment as Lambda runs on the
  container. Here, as FFmpeg is a third party software, diving deeper
  into the dependency chain and verifying the version compatibilities is
  very hard to do. Unfortunately going further, this is bound to go into
  architecture and code support which is out of AWS Support scope 1. I
  hope you understand our limitations. However should FFmpeg support
  have any questions specific to the Lambda platform, please do let us
  know and we will be happy to assist. We will be in better position to
  investigate further once you receive an update from the FFmpeg support
  suggesting an issue from Lambda end.

  Upon AWS suggestion, I contacted FFMPEG on the developers mailing list, my message was rejected with the reason being that its more suited to ffmpeg users mailing list than developers. I sent an email to ’ffmpeg-user@ffmpeg.org’ a week ago and did not get any response yet.

  I then went and built a dynamically linked ffmpeg version making sure to package all libraries, checked ddl on each one, then made a small lambda function that looped over all binaries and ddled each one of them, compared that to the output I got from Amazon Linux and the same dependencies/versions exists on both lambda and the AWS Linux instance yet ffmpeg still fails on lambda.

  You can find a detailed log file here : https://www.datafilehost.com/d/6e5e21bb

  And this is a sample of the errors I’m getting, repeated across the entire log file :

  2018-08-14T12:27:10.874Z [h264 @ 0x65c2fc0] concealing 2628 DC, 2628
  AC, 2628 MV errors in P frame

  2018-08-14T12:27:10.874Z [aac @ 0x65d2f00] channel element 2.11 is not
  allocated

  2018-08-14T12:27:10.874Z Error while decoding stream #0:1 : Invalid
  data found when processing input

  2018-08-14T12:27:10.874Z [h264 @ 0x67e86c0] Invalid NAL unit size
  (108085662 > 1649).

  2018-08-14T12:27:10.874Z [h264 @ 0x67e86c0] Error splitting the input
  into NAL units.

  2018-08-14T12:27:10.874Z [aac @ 0x65d2f00] channel element 2.0 is not
  allocated

  2018-08-14T12:27:10.874Z Error while decoding stream #0:1 : Invalid
  data found when processing input

  2018-08-14T12:27:10.874Z [h264 @ 0x68189c0] Invalid NAL unit size
  (71106974 > 1085).

  2018-08-14T12:27:10.874Z [h264 @ 0x68189c0] Error splitting the input
  into NAL units.

  2018-08-14T12:27:10.874Z [aac @ 0x65d2f00] Pulse tool not allowed in
  eight short sequence.

  This log is generated when trying to perform an HLS transcoding on this file : https://www.datafilehost.com/d/999a4492

  Note that the issue is not related to that file alone nor is it related to HLS, its general and happen on all videos and any ffmpeg command that tries to seek the stream, even tried extracting a single frame from a video using the simplest form possible for example : ffmpeg -ss 00:00:02 -I file.mp4 -vframes 1 -y output.jpg also fails with the same errors in the log file.

  Not sure how to debug this further. Tried enabling debug logs with ‘-loglevel debug’ but did not give me any extra info. Any help or suggestions

• 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.