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• FFMPEG concatinate audio file and add background music

  2 mai 2018, par Swetanka Jaiswal

  I want to concatenate 2 audio files and play the third audio in the background. I used below code

  ffmpeg -i 1.mp3 -i 2.mp3 -i background.mp3 -filter_complex "[0:0][1:0]concat=n=2:v=0:a=1,volume=1dB[a0] ;[2]volume=0.5dB[a1] ;[a0][a1]amerge[a]" -map "[a]" -strict -2 -y final.mp3

  suggested here ffmpeg : How to concat audio files and add background music in a single command ?

  But it is giving error "The following filters could not choose their formats : Parsed_amerge_3 Consider inserting the (a)format filter near their input or output."

  Please let me know what I’m doing wrong.

• FFMpeg - add background music

  22 août 2018, par jacky brown

  here is what i have :
  input1.avi - video that contain sounds.
  input2.avi - video that doesn’t contain sounds.
  music.mp3 - audio file.

  i want to add background music(music.mp3 file) to the video.

  C:\input1.avi -i C:\music.mp3 -shortest -c:v copy -c:a copy C:\output1.avi

  then output1.avi is the same as input1 - movie with sounds but without the background music (music.mp3)

  when i try to use the other file (video without sounds) :

  C:\input2.avi -i C:\music.mp3 -shortest -c:v copy -c:a copy C:\output2.avi

  then output2.avi is the same as input2 + it have the background music.

  so why input1 does not contain the background music ???
  and how can i decrease or increase the volume of music.mp3 file ?

  thanks.

  ---

  console output :

  C:\motionbee\ffmpeg\bin>ffmpeg -i C:\input.avi
  
  -i C:\music.mp3 -shortest -c:v copy -filter_
  
  complex "[0:a]aformat=fltp:44100:stereo[0a];[1]aformat=fltp:44100:stereo,volume=
  
  1.5[1a];[0a][1a]amix[a]" -map 0:v -map "[a]" -ac 2 C:\output1.avi
  
  ffmpeg version N-78949-g6f5048f Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 5.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
  
  isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
  
  le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
  
  enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
  
  ibilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopenc
  
  ore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --ena
  
  ble-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable
  
  -libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --ena
  
  ble-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx
  
  264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable
  
  -lzma --enable-decklink --enable-zlib
  
    libavutil      55. 19.100 / 55. 19.100
  
    libavcodec     57. 27.101 / 57. 27.101
  
    libavformat    57. 28.100 / 57. 28.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 39.100 /  6. 39.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, avi, from 'C:\input.avi':
  
    Metadata:
  
      encoder         : Lavf57.28.100
  
    Duration: 00:02:05.76, start: 0.000000, bitrate: 450 kb/s
  
      Stream #0:0: Video: mpeg4 (Simple Profile) (XVID / 0x44495658), yuv420p, 720
  
  x480 [SAR 1:1 DAR 3:2], 440 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
  
      Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, stereo, s16p, 128 k
  
  b/s
  
  [mp3 @ 00000000005c8020] Skipping 0 bytes of junk at 32370.
  
  Input #1, mp3, from 'C:\music.mp3':
  
    Metadata:
  
      title           : Broadcast News Package - News Intro
  
      artist          : After Effects News Template
  
    Duration: 00:01:57.89, start: 0.025057, bitrate: 194 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc56.26
  
  File 'C:\output1.avi' already exists. Overwrite
  
  ? [y/N] y
  
  Output #0, avi, to 'C:\output1.avi':
  
    Metadata:
  
      ISFT            : Lavf57.28.100
  
      Stream #0:0: Video: mpeg4 (XVID / 0x44495658), yuv420p, 720x480 [SAR 1:1 DAR
  
   3:2], q=2-31, 440 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
  
      Stream #0:1: Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 44100 Hz, stereo
  
  , fltp (default)
  
      Metadata:
  
        encoder         : Lavc57.27.101 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 (mp3) -> aformat
  
    Stream #1:0 (mp3) -> aformat
  
    Stream #0:0 -> #0:0 (copy)
  
    amix -> Stream #0:1 (libmp3lame)
  
  Press [q] to stop, [?] for help
  
  frame= 3118 fps=0.0 q=-1.0 Lsize=    6917kB time=00:02:05.76 bitrate= 450.6kbits
  
  /s speed= 867x
  
  video:6754kB audio:74kB subtitle:0kB other streams:0kB global headers:0kB muxing
  
   overhead: 1.307944%

• Extract raw audio frames from OGG music file with Android NDK

  31 octobre 2018, par thenaoh

  In my Android app, I would like to be able to process audio on the fly from an OGG file by extracting audio samples, process them and redirect them to the audio output.

  I know how to make the last 2 steps using Android NDK, but I don’t know how to extract audio samples to get them in an array of floats or shorts.

  I tried to make this code work that, apparently, can extract raw audio samples on the fly.

  The problem is : I don’t manage to add FFMpeg in my project. I tried many tutorials (like this one), but it seems pretty difficult since I work on Windows. So after a while, I found Prebuild FFMpeg for Android, that seems interesting since it’s available for armeabi-v7a, arm64-v8a, x86 and x86_64 architectures, but again, I don’t understand how to add it in my project.

  I also took a look at libogg, libvorbis and vorbisfile, but I have no idea how to add them in my project.

  So, does anyone have a working example on how to extract audio samples from an OGG file on the fly ?

  Thanks for your help.