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• Merge Audio with Video is not working probably using android 9 & 8

  22 juillet 2021, par ebdaa app

  I am trying to use the following code to merge audio with video,

  


      cmd = "-stream_loop -1 -i " + videoUri + " -i " + audioPath + " -shortest -map 0:v:0 -map 1:a:0 -y " + videoOutputPath;

    long executionId = FFmpeg.executeAsync(cmd, new ExecuteCallback() {

        @Override
        public void apply(final long executionId, final int returnCode) {
            if (returnCode == RETURN_CODE_SUCCESS) {
                playVedio(videoOutputPath);
            } else {
                ErrorLogger();
            }
        }
    });


  


  as you can see the above code does the following things :

  


  

  1. replace audio in video with new one.

  


  2. loop the video until the new audio ends.

  


  


  everything works perfectly when trying to run the code using both android 11 and 10 , however , when try to use android 9 or 8 , the first and the last 2 seconds of the new audio will be trimmed in the generated video and I am not able to know why

  


  I am using 4.4 version of the mobile-ffmpeg

  


   com.arthenica:mobile-ffmpeg-full:4.4


  


  please find the log from android 8/9

  


   I/mobile-ffmpeg: ffmpeg version v4.4-dev-416
      Copyright (c) 2000-2020 the FFmpeg developers
       built with Android (6454773 based on r365631c2) clang version 9.0.8 (https://android.googlesource.com/toolchain/llvm-project 98c855489587874b2a325e7a516b99d838599c6f) (based on LLVM 9.0.8svn)
       configuration: --cross-prefix=i686-linux-android- --sysroot=/files/android-sdk/ndk/21.3.6528147/toolchains/llvm/prebuilt/linux-x86_64/sysroot --prefix=/home/taner/Projects/mobile-ffmpeg/prebuilt/android-x86/ffmpeg --pkg-config=/usr/bin/pkg-config --enable-version3 --arch=i686 --cpu=i686 --cc=i686-linux-android24-clang --cxx=i686-linux-android24-clang++ --extra-libs='-L/home/taner/Projects/mobile-ffmpeg/prebuilt/android-x86/cpu-features/lib -lndk_compat' --target-os=android --disable-neon --disable-asm --disable-inline-asm --enable-cross-compile --enable-pic --enable-jni --enable-optimizations --enable-swscale --enable-shared --enable-v4l2-m2m --disable-outdev=fbdev --disable-indev=fbdev --enable-small --disable-openssl --disable-xmm-clobber-test --disable-debug --enable-lto --disable-neon-clobber-test --disable-programs --disable-postproc --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-static --disable-sndio --disable-schannel --disable-securetransport --disable-xlib --disable-cuda --disable-cuvid --disable-nvenc --disable-vaapi --disable-vdpau --disable-videotoolbox --disable-audiotoolbox --disable-appkit --disable-alsa --disable-cuda --disable-cuvid --disable-nvenc --disable-vaapi --disable-vdpau --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-gmp --enable-gnutls --enable-libmp3lame --enable-libass --enable-iconv --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libxml2 --enable-libopencore-amrnb --enable-libshine --enable-libspeex --enable-libwavpack --enable-libkvazaar --enable-libilbc --enable-libopus --enable-libsnappy --enable-libsoxr --enable-libaom --enable-libtwolame --disable-sdl2 --enable-libvo-amrwbenc --enable-zlib --enable-mediacodec
       libavutil      56. 55.100 / 56. 55.100
       libavcodec     58. 96.100 / 58. 96.100
       libavformat    58. 48.100 / 58. 48.100
       libavdevice    58. 11.101 / 58. 11.101
       libavfilter     7. 87.100 /  7. 87.100
       libswscale      5.  8.100 /  5.  8.100
       libswresample   3.  8.100 /  3.  8.100
 I/mobile-ffmpeg: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/Download/Videoes/share_video_2.mp4':
       Metadata:
         major_brand     : 
     isom
         minor_version   : 
     512
         compatible_brands: 
     isomiso2avc1mp41
         encoder         : 
 I/mobile-ffmpeg: Lavf58.44.100
       Duration: 
     00:00:08.21
     , start: 
     0.000000
     , bitrate: 
     477 kb/s
         Stream #0:0
     (und)
     : Video: h264 (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 473 kb/s
     , 
     29.97 fps, 
     29.97 tbr, 
     30k tbn, 
     59.94 tbc
      (default)
         Metadata:
           handler_name    : 
     VideoHandler
 I/mobile-ffmpeg: Input #1, mp3, from '/storage/emulated/0/Download/001001.mp3':
 I/mobile-ffmpeg:   Metadata:
         album           : 
     Mishary Alafasi Musshaf
         artist          : 
     Mishary Alafasi
         comment         : 
     www.mp3quran.net
         genre           : 
     Quran
         title           : 
     Al-Fatihah
         date            : 
     2007
         encoder         : 
     Lavf58.48.100
       Duration: 
     00:00:12.41
     , start: 
     0.011995
     , bitrate: 
     132 kb/s
         Stream #1:0
 I/mobile-ffmpeg: : Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s
         Metadata:
           encoder         : 
     Lavf
         Stream #1:1
     : Video: png, pal8(pc), 200x159 [SAR 2835:2835 DAR 200:159]
     , 
     90k tbr, 
     90k tbn, 
     90k tbc
      (attached pic)
         Metadata:
           comment         : 
     Cover (front)
 I/mobile-ffmpeg: Stream mapping:
       Stream #0:0 -> #0:0
      (h264 (native) -> mpeg4 (native))
       Stream #1:0 -> #0:1
      (mp3 (mp3float) -> aac (native))
     Press [q] to stop, [?] for help
 I/mobile-ffmpeg: frame=    0 fps=0.0 q=0.0 size=       0kB time=-577014:32:22.77 bitrate=  -0.0kbits/s speed=N/A    
 W/mobile-ffmpeg: [graph 0 input from stream 0:0 @ 0xd429d2e0] sws_param option is deprecated and ignored
 D/EGL_emulation: eglMakeCurrent: 0xf08c6ee0: ver 2 0 (tinfo 0xf08d2810)
 I/mobile-ffmpeg: Output #0, mp4, to '/storage/emulated/0/Download/v001001.mp4':
       Metadata:
         major_brand     : 
     isom
         minor_version   : 
 I/mobile-ffmpeg: 512
         compatible_brands: 
     isomiso2avc1mp41
         encoder         : 
     Lavf58.48.100
         Stream #0:0
     (und)
     : Video: mpeg4 (mp4v / 0x7634706D), yuv420p(progressive), 640x360 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s
     , 
     29.97 fps, 
     30k tbn, 
     29.97 tbc
      (default)
         Metadata:
           handler_name    : 
     VideoHandler
           encoder         : 
     Lavc58.96.100 mpeg4
         Side data:
 I/mobile-ffmpeg:       
     cpb: 
     bitrate max/min/avg: 0/0/200000 buffer size: 0 
     vbv_delay: N/A
 I/mobile-ffmpeg:     Stream #0:1
     : Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
         Metadata:
           encoder         : 
     Lavc58.96.100 aac
 I/mobile-ffmpeg: frame=   83 fps=0.0 q=30.2 size=       0kB time=00:00:02.78 bitrate=   0.1kbits/s speed=5.54x    
 I/mobile-ffmpeg: frame=  199 fps=198 q=31.0 size=     256kB time=00:00:06.64 bitrate= 315.8kbits/s speed=6.61x    
 I/mobile-ffmpeg: frame=  300 fps=199 q=31.0 size=     512kB time=00:00:10.00 bitrate= 419.1kbits/s speed=6.65x    
 I/mobile-ffmpeg: frame=  372 fps=204 q=31.0 Lsize=     733kB time=00:00:12.39 bitrate= 484.5kbits/s speed= 6.8x    
     video:526kB audio:195kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 
     1.630239%
 I/mobile-ffmpeg: [aac @ 0xd431fc00] Qavg: 164.571


  


• Can ffmpeg copy metadata/ID3 from FLAC to MP3 ?

  9 avril 2019, par krypterro

  I have a bit of Python code that loops through audio files, finds .FLAC files, and then uses the Python subcommand to run ffmpeg. It works. The audio is fine, but even though I see the metadata in the shell, it doesn’t transfer the data to the ID3 tags in the MP3, and I am using the example found in the previous post here. Here’s the command :

  cmd = 'ffmpeg -y -i "' + src + '" -codec:a libmp3lame -q:a 0 -map_metadata 0 -id3v2_version 3 -write_id3v1 1 "' + dst + '"'

  Which works out to :

  ffmpeg -y -i "source.flac" -codec:a libmp3lame -q:a 0 -map_metadata 0 -id3v2_version 3 -write_id3v1 1 "destination.mp3"

  And here is the log dump :

  /usr/local/bin/python3.7 /home/krypterro/PycharmProjects/mediaman/RipFLAC.py
  
  1 Music Files Found
  
  2019-04-09 14:32:47.758 | INFO     | __main__:main:31 - Start of program
  
  2019-04-09 14:32:48.110 | DEBUG    | __main__:ripmp3:206 - Running Command: ffmpeg -y -i "/home/krypterro/audio/music_in/Visions/01-grimes-laughing_and_not_being_normal.flac" -acodec libmp3lame -ab 192000 "/home/krypterro/audio/music_out/Visions/01-grimes-laughing_and_not_being_normal.mp3"
  
  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
  
    WARNING: library configuration mismatch
  
    avcodec     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 --enable-version3 --disable-doc --disable-programs --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract --enable-libvo_amrwbenc
  
    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, flac, from '/home/krypterro/audio/music_in/Visions/01-grimes-laughing_and_not_being_normal.flac':
  
    Metadata:
  
      ARTIST          : Grimes
  
      TITLE           : Laughing And Not Being Normal
  
      ALBUM           : Art Angels
  
      DATE            : 2015
  
      track           : 1
  
      GENRE           : Electronic
  
      disc            : 1
  
      TOTALDISCS      : 1
  
      TOTALTRACKS     : 15
  
      LANGUAGE        : English
  
      RIP DATE        : 2015-12-12
  
      RETAIL DATE     : 2015-00-00
  
      MEDIA           : CD
  
      ENCODER         : FLAC 1.2.1 -8 -V
  
      RIPPING TOOL    : EAC 1.0 Beta 3
  
      RELEASE TYPE    : Retail
  
      ORGANIZATION    : 4AD
  
      CATALOG         : CAD3535CD
  
    Duration: 00:01:47.51, start: 0.000000, bitrate: 743 kb/s
  
      Stream #0:0: Audio: flac, 44100 Hz, stereo, s16
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (flac (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  Output #0, mp3, to '/home/krypterro/audio/music_out/Visions/01-grimes-laughing_and_not_being_normal.mp3':
  
    Metadata:
  
      TPE1            : Grimes
  
      TIT2            : Laughing And Not Being Normal
  
      TALB            : Art Angels
  
      TDRC            : 2015
  
      TRCK            : 1
  
      TCON            : Electronic
  
      TPOS            : 1
  
      TOTALDISCS      : 1
  
      TOTALTRACKS     : 15
  
      TLAN            : English
  
      RIP DATE        : 2015-12-12
  
      RETAIL DATE     : 2015-00-00
  
      MEDIA           : CD
  
      CATALOG         : CAD3535CD
  
      RIPPING TOOL    : EAC 1.0 Beta 3
  
      RELEASE TYPE    : Retail
  
      ORGANIZATION    : 4AD
  
      TSSE            : Lavf57.83.100
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc57.107.100 libmp3lame
  
  size=    2522kB time=00:01:47.52 bitrate= 192.1kbits/s speed=  41x    
  
  video:0kB audio:2521kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.042501%
  
  2019-04-09 14:32:50.811 | INFO     | __main__:main:73 - End of program
  
  
  
  Process finished with exit code 0

• MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?

  23 janvier 2019, par AR5

  I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
  The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)

  Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.

  I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?

  I have already tried

  I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.

  I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.

  I also checked if both files were being served as 206 Partial Content and they both are indeed.

  I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv

  I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.

  I am really stuck on this issue, any help will be really appreciated.

  ---

  Edit

  I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :

  avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"

  Here is the command and log output from new server :

  ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
  
  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 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
  
  Input #0, mp3, from '/test/Track 01.mp3':
  
    Metadata:
  
      album           : Future Hndrxx Presents: The WIZRD
  
      artist          : Future
  
      genre           : Hip-Hop
  
      title           : Never Stop
  
      track           : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      date            : 2019
  
      encoder         : Lavf56.40.101
  
    Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60
  
      Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  [mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
  
  Please consider specifying a lower framerate, a different muxer or -vsync 2
  
  Output #0, mp3, to '/test/Track 01 (converted).mp3':
  
    Metadata:
  
      TALB            : Future Hndrxx Presents: The WIZRD
  
      TPE1            : Future
  
      TCON            : Hip-Hop
  
      TIT2            : Never Stop
  
      TRCK            : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      TDRC            : 2019
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
        encoder         : Lavc56.60.100 png
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (png (native) -> png (native))
  
    Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
  
  frame=    1 fps=0.1 q=-0.0 Lsize=    4788kB time=00:04:51.39 bitrate= 134.6kbits/s
  
  video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%

  Samples of MP3 files

  I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing