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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 No encoder found for codec id 8

  3 septembre 2018, par raspiboele

  I’m using a Raspberry Pi to restreem a (https) .m3u8 stream, to a local MJPEG-stream with FFMPEG.
  Becaus I have a Nest Cam / Nest Hello and I want to import that stream into my Fibaro Homecenter (Smarthome Basestation), but the Fibaro Homecenter only eats the MJPEG format. So I made a Raspberry Pi with FFMPEG to do the trick.

  This is my /etc/ffserver.conf file :

  HTTPPort 8090
  
  
  
  HTTPBindAddress 0.0.0.0
  
  
  
  MaxHTTPConnections 2000
  
  
  
  MaxClients 1000
  
  
  
  MaxBandwidth 1000
  
  
  
  CustomLog -
  
  
  
  <feed>
  
  File /tmp/feed1.ffm
  
  FileMaxSize 30M
  
  </feed>
  
  
  
  <stream>
  
  Feed feed1.ffm
  
  Format mpjpeg
  
  VideoFrameRate 2
  
  VideoIntraOnly
  
  NoAudio
  
  Strict -1
  
  </stream>

  My command :

   ffmpeg -i "https://path-to-stream.com/chucklist.m3u8" http://localhost:8090/feed1.ffm

  Output :

  ffmpeg version N-89723-g2ca65fc7b7 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1+deb9u1) 20170516
  
    configuration: --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-gnutls
  
    libavutil      56.  7.100 / 56.  7.100
  
    libavcodec     58.  9.100 / 58.  9.100
  
    libavformat    58.  3.100 / 58.  3.100
  
    libavdevice    58.  0.100 / 58.  0.100
  
    libavfilter     7. 11.101 /  7. 11.101
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  [hls,applehttp @ 0x1d16e70] Opening 'https://path-to-stream.com/chucklist.ts' for reading
  
  [hls,applehttp @ 0x1d16e70] Opening 'https://path-to-stream.com/chucklist.ts' for reading
  
  Input #0, hls,applehttp, from 'https://path-to-stream.com/chucklist.m3u8':
  
    Duration: N/A, start: 38789.189067, bitrate: N/A
  
    Program 0
  
      Metadata:
  
        variant_bitrate : 0
  
      Stream #0:0: Data: timed_id3 (ID3  / 0x20334449)
  
      Metadata:
  
        variant_bitrate : 0
  
      Stream #0:1: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuvj420p(pc, bt709), 1152x864 [SAR 1:1 DAR 4:3], 15 tbr, 90k tbn, 30 tbc
  
      Metadata:
  
        variant_bitrate : 0
  
      Stream #0:2: Audio: aac (LC) ([15][0][0][0] / 0x000F), 16000 Hz, mono, fltp
  
      Metadata:
  
        variant_bitrate : 0
  
  [tcp @ 0x27b36f0] Connection to tcp://localhost:8090 failed (Connection refused), trying next address
  
  [ffm @ 0x237a480] no encoder found for codec id 8
  
  http://localhost:8090/feed1.ffm: Invalid argument

  As you can see, I get an error : no encoder found for codec id 8
  It is frutstrating me, because I’m looking for a solution for three days now. I can’t even find a list of encoders and the coresponding ID’s. Do I have to enable maybe something in the ./configure ?

  My target is to get a stream at : http://localhost:8090/test1.mjpg

  Can anyone help me please ?

• generating silence audio using ffmpeg with specific bitrate and duration time less than one second cannot get the audio with the desired bitrate

  28 août 2018, par bambooom

  I’d like to generate a silence audio file using ffmpeg with specific bitrate and the duration is less than one second. I found that the generated audio file does not have the desired bitrate. But if I expand the duration to 2 seconds, the bitrate seems to be correct.

  For example :

  ffmpeg -f lavfi -i anullsrc=r=22050:cl=mono -t 0.3 -acodec mp3 -ab 48k -y silence.mp3

  The generated silence.mp3’s bitrate seems not 48kb/s :

  $ ffprobe silence.mp3                                                                                           
  
  ffprobe version 3.4.2 Copyright (c) 2007-2018 the FFmpeg developers
  
    built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --disable-jack --enable-gpl --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
  
    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, mp3, from 'silence.mp3':
  
    Metadata:
  
      encoder         : Lavf57.83.100
  
    Duration: 00:00:00.37, start: 0.050113, bitrate: 52 kb/s
  
      Stream #0:0: Audio: mp3, 22050 Hz, mono, s16p, 48 kb/s

  If the duration is 2 seconds, bitrate seems OK :

  $ ffmpeg -f lavfi -i anullsrc=r=22050:cl=mono -t 2 -acodec mp3 -ab 48k -y silence.mp3
  
  $ ffprobe silence.mp3                                                                                           
  
  ffprobe version 3.4.2 Copyright (c) 2007-2018 the FFmpeg developers
  
    built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --disable-jack --enable-gpl --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
  
    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, mp3, from 'silence.mp3':
  
    Metadata:
  
      encoder         : Lavf57.83.100
  
    Duration: 00:00:02.06, start: 0.050113, bitrate: 48 kb/s
  
      Stream #0:0: Audio: mp3, 22050 Hz, mono, s16p, 48 kb/s

  Is there any way to keep the bitrate as desired ?