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• Unable to convert .mp3 to .m4a using ffmpeg [closed]

  26 décembre 2020, par rahulg

  I am fully aware the legal constraints in using libfaac but this is just for my testing purpose.

  



  I have compiled ffmpeg with faac enabled. So when I tried to convert an .mp3 to a .m4a here is the error that I am getting. Please provide a resolution to this problem. I tried it on two different sources of .mp3, still I am getting the same error.

  



  [user@ip-10-161-13-26 ~]$ ffmpeg  -i Kalimba.mp3 -c:a libfaac Kalimba.m4a
ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
  built on May  4 2013 09:33:27 with gcc 4.4.6 20120305 (Red Hat 4.4.6-4)
  configuration: --enable-libfaac --enable-nonfree --disable-yasm
  libavutil      51. 54.100 / 51. 54.100
  libavcodec     54. 23.100 / 54. 23.100
  libavformat    54.  6.100 / 54.  6.100
  libavdevice    54.  0.100 / 54.  0.100
  libavfilter     2. 77.100 /  2. 77.100
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
[mp3 @ 0x2464680] Header missing
[mp3 @ 0x2463100] max_analyze_duration 5000000 reached at 5015510
[mp3 @ 0x2463100] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'Kalimba.mp3':
  Metadata:
    publisher       : Ninja Tune
    track           : 1
    album           : Ninja Tuna
    artist          : Mr. Scruff
    album_artist    : Mr. Scruff
    title           : Kalimba
    genre           : Electronic
    composer        : A. Carthy and A. Kingslow
    date            : 2008
  Duration: 00:05:50.60, start: 0.000000, bitrate: 191 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s
    Stream #0:1: Video: mjpeg, yuvj420p, 512x512, 90k tbr, 90k tbn, 90k tbc
    Metadata:
      title           : thumbnail
      comment         : Cover (front)
Output #0, ipod, to 'Kalimba.m4a':
  Metadata:
    publisher       : Ninja Tune
    track           : 1
    album           : Ninja Tuna
    artist          : Mr. Scruff
    album_artist    : Mr. Scruff
    title           : Kalimba
    genre           : Electronic
    composer        : A. Carthy and A. Kingslow
    date            : 2008
    Stream #0:0: Video: none, q=2-31, 128 kb/s, 90k tbn
    Metadata:
      title           : thumbnail
      comment         : Cover (front)
    Stream #0:1: Audio: none, 0 channels, 128 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 (mjpeg -> ?)
  Stream #0:0 -> #0:1 (mp3 -> libfaac)
Encoder (codec none) not found for output stream #0:0


  



  MP3 file is at http://db.tt/HtpEBpFU

  



  Also while using Faac independently I get this weird error for any file.

  



  Freeware Advanced Audio Coder
FAAC 1.28

Couldn't open input file sample.mp3


  


• Unable to convert .mp3 to .m4a using ffmpeg [closed]

  4 mai 2013, par rahulg

  I am fully aware the legal constraints in using libfaac but this is just for my testing purpose.

  I have compiled ffmpeg with faac enabled. So when I tried to convert an .mp3 to a .m4a here is the error that I am getting. Please provide a resolution to this problem. I tried it on two different sources of .mp3, still I am getting the same error.

  [user@ip-10-161-13-26 ~]$ ffmpeg  -i Kalimba.mp3 -c:a libfaac Kalimba.m4a
  
  ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
  
    built on May  4 2013 09:33:27 with gcc 4.4.6 20120305 (Red Hat 4.4.6-4)
  
    configuration: --enable-libfaac --enable-nonfree --disable-yasm
  
    libavutil      51. 54.100 / 51. 54.100
  
    libavcodec     54. 23.100 / 54. 23.100
  
    libavformat    54.  6.100 / 54.  6.100
  
    libavdevice    54.  0.100 / 54.  0.100
  
    libavfilter     2. 77.100 /  2. 77.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0. 15.100 /  0. 15.100
  
  [mp3 @ 0x2464680] Header missing
  
  [mp3 @ 0x2463100] max_analyze_duration 5000000 reached at 5015510
  
  [mp3 @ 0x2463100] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from 'Kalimba.mp3':
  
    Metadata:
  
      publisher       : Ninja Tune
  
      track           : 1
  
      album           : Ninja Tuna
  
      artist          : Mr. Scruff
  
      album_artist    : Mr. Scruff
  
      title           : Kalimba
  
      genre           : Electronic
  
      composer        : A. Carthy and A. Kingslow
  
      date            : 2008
  
    Duration: 00:05:50.60, start: 0.000000, bitrate: 191 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s
  
      Stream #0:1: Video: mjpeg, yuvj420p, 512x512, 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        title           : thumbnail
  
        comment         : Cover (front)
  
  Output #0, ipod, to 'Kalimba.m4a':
  
    Metadata:
  
      publisher       : Ninja Tune
  
      track           : 1
  
      album           : Ninja Tuna
  
      artist          : Mr. Scruff
  
      album_artist    : Mr. Scruff
  
      title           : Kalimba
  
      genre           : Electronic
  
      composer        : A. Carthy and A. Kingslow
  
      date            : 2008
  
      Stream #0:0: Video: none, q=2-31, 128 kb/s, 90k tbn
  
      Metadata:
  
        title           : thumbnail
  
        comment         : Cover (front)
  
      Stream #0:1: Audio: none, 0 channels, 128 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (mjpeg -> ?)
  
    Stream #0:0 -> #0:1 (mp3 -> libfaac)
  
  Encoder (codec none) not found for output stream #0:0

  MP3 file is at http://db.tt/HtpEBpFU

  Also while using Faac independently I get this weird error for any file.

  Freeware Advanced Audio Coder
  
  FAAC 1.28
  
  
  
  Couldn't open input file sample.mp3

• ffmpeg : concatinating files creates audio artefacts

  28 octobre 2022, par LML

  I'm currently trying to create a video out of multiple short video files. However, the final video always has audio artefacts, where it sounds like a short high pitch or echo at certain times during the audio. All the audio is a text-to-speech generated voice. No music. The artefacts appear sometimes more, sometimes less. But I would obviously prefer to have 0 of it.

  


  My starting point is a long audio file (mono with audio codec "mp3" according to ffprobe). Within that file are a bunch of short pauses of 4-5 seconds. I detect the silences and create individual audio files from there. Afterwards I create an mp4 file with this audio and a still image. Up to this point, the audio is perfectly fine and sounds the exact same as in the original file.

  


  After this I want to create the final video : each of the individual parts added into one long video. There is a transition between each file to mark the changing of image and audio. But even when skipping the transition and simply adding all of these clips that were generated the same way together, the artefacts are still present.

  


  The commands I use to create the different files.

  


  Create individual audio files :
.\ffmpeg.exe -y -hide_banner -i TTSAudio.mp3 -ss 359.944 -to 372.02479 -c copy partXY.mp3

  


  Create individual video files by using a .png file as the video stream and the partXY.mp3 as the audio stream :
.\ffmpeg.exe -y -hide_banner -framerate 30 -loop 1 -i XY_full.png -i partXY.mp3 -c:v libx265 -c:a copy -shortest partXY.mp4

  


  For concatenating the files :
.\ffmpeg.exe -y -hide_banner -i part000.mp4 -i part001.mp4 -i part002.mp4 -filter_complex "[0:v] [0:a] [1:v] [1:a] [2:v] [2:a] concat=n=3:v=1:a=1 [v] [a]" -map "[v]" -map "[a]" -c:v copy -c:a copy final_video.mp4

  


  I've tried a lot of different things and codecs for the audio, without any luck. I use h265, as using h264 was causing weird video artefacts after uploading the file to YouTube.
I have tried reencoding, instead of copying (-c:a copy) at various stages, especially the final video. All without any luck.
I've used the different concatenation where you provide a list of files, which created a whole different set of problems.

  


  I've managed to filter the artefacts out by using -af "lowpass=f=2800", but that changes the voice a lot. I was also not able to notice the pitch visually when opening the audio in audacity, for example.

  


  Example :
https://soundcloud.com/thelml/sets/ffmpeg-audio-artefacts/s-LNr6UaMPgz9?si=f7b30e1e64bf4333ad055fa1fe21e9ec
Due to the files being so short, I seem to have to sometimes have to replay the bugged file to hear the artefact.

  


  So my question is : how do I fix this, without using a lowpass that basically changes the whole voice ?