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• ffmpeg amix filter on mp3 with image

  20 November 2018, by cutoffurmind

  I use ffmpeg to concatenate 2 mp3 files with amix filter (see full log).

  It works fine for mp3 files without any meta but it fails with mp3 files with cover image in meta, result file length is less then 1 second.

  How could I fix it in same command?

  Here is full log:

  localhost:Music user$ ffmpeg -i input.mp3 -i /opt/docker/tag_long.mp3 -filter_complex amix=inputs=2:duration=shortest,volume=2 -codec:a libmp3lame -q:a 5  out.mp3 -report
  
  ffmpeg started on 2018-11-11 at 13:19:50
  
  Report written to "ffmpeg-20181111-131950.log"
  
  ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gpl --enable-libmp3lame --enable-libopus --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-opencl --enable-videotoolbox
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  [mp3 @ 0x7fe506000000] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from 'input.mp3':
  
    Metadata:
  
      artist          : Paul
  
      album           : Underground Vol. 17
  
      title           : Crazy
  
      track           : 11/20
  
      date            : 2017
  
    Duration: 00:04:46.23, start: 0.000000, bitrate: 324 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
  
      Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 500x500 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  [mp3 @ 0x7fe50601aa00] Estimating duration from bitrate, this may be inaccurate
  
  Input #1, mp3, from '/opt/docker/tag_long.mp3':
  
    Metadata:
  
      genre           : Blues
  
      id3v2_priv.XMP  : &lt;?xpacket begin="\xef\xbb\xbf" id="W5M0MpCehiHzreSzNTczkc9d"?>\x0a\x0a \x0a  <rdf 128="128" kb="kb"></rdf>s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s
  
  File 'out.mp3' already exists. Overwrite ? [y/N] Y
  
  Stream mapping:
  
    Stream #0:0 (mp3float) -> amix:input0 (graph 0)
  
    Stream #1:0 (mp3float) -> amix:input1 (graph 0)
  
    volume (graph 0) -> Stream #0:0 (libmp3lame)
  
    Stream #0:1 -> #0:1 (mjpeg (native) -> png (native))
  
  Press [q] to stop, [?] for help
  
  [swscaler @ 0x7fe506045000] deprecated pixel format used, make sure you did set range correctly
  
  [mp3 @ 0x7fe507810000] 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 'out.mp3':
  
    Metadata:
  
      TPE1            : Paul
  
      TALB            : Underground Vol. 17
  
      TIT2            : Crazy
  
      TRCK            : 11/20
  
      TDRC            : 2017
  
      TSSE            : Lavf58.20.100
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp (default)
  
      Metadata:
  
        encoder         : Lavc58.35.100 libmp3lame
  
      Stream #0:1: Video: png, rgb24(progressive), 500x500 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
        encoder         : Lavc58.35.100 png
  
  frame=    1 fps=0.0 q=-0.0 Lsize=     496kB time=00:00:00.26 bitrate=15501.4kbits/s speed=0.336x    
  
  video:495kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.084876%

• Heroku django weird tempfile behavior

  13 February 2019, by BenFire

  I’m writing a django application, where, a user records a message, the message is converted with FFMPEG and uploaded to a FS,now, FFMPEG doesn’t allow file overwrite, so i’m creating a new file and than replacing the old one before inserting it my filesystem. so basically:

  The file is recorded in the front-end -> sent to the backend and stored as a tempfile (automatically by django) -> the files gets converted -> the file is uploaded to the fs.

  (the file is converted from around 2mb to 200kb, which is an indicator to if the file was converted)

  now, to replace the original file, I use shutil.move(), and when i run it on my system, and then I look in my fs, the file sure is only 200kb, but when i push the same code to Heroku, i do the same operation, when look in the fs I see that the new file is still 2mb, also, I know the conversion was done from the server logs, whitch means that the file replacing failed but only on Heroku, am i missing something?

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

  23 January 2019, by 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