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• concat 2 mini audio files and make a loop and add a background music

  11 octobre 2020, par Johny Sharma

  I Need to concatenate 2 small audio files with loop and add background music in a single command.

  



  I am capable to concatenate two audio files with a background music. My above given code is working.

  



  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,aformat=fltp, pan=stereo|c0=c0|c1=c0[a0]; 
[2]volume=0.5dB,aformat=fltp,pan=stereo|c0=c0|c1=c1[a1];[a0][a1]amix=inputs=2:duration=longest,aformat=fltp[a]"
-map "[a]" -strict -2 -y output.mp3


  



  but i want to a make a loop of the concatenated files till the end of the background music. background music is longer than approx 5 times from concatenated files.

  



  If someone can suggest a single command solution.

  



  I know about amovie tag but unfortunately its not possible to use in here because amovie requires file name which is not possible with concatenated files as per my knowledge.

  



  Can anyone help me how can i achieve my goal !

  



  Thanks

  


• add part of audio with loop to video using ffmpeg

  7 janvier 2019, par 1234567

  I have a video 5 min 30 seconds long and an audio 4 mins 30 seconds long

  I want to add part of audio(part of audio from 1 min 25 sec to 1 min 47 sec) to video(part of video from 2 min 30 sec to 3 min 55 sec)

  what I have been able to do is loop audio and add it to video

  with this command

  "-y","-i",j, "-filter_complex",
  
                          "amovie="+audio+":loop=1000," +
  
  "asetpts=N/SR/TB,atrim=0:85,adelay=150000|150000,apad," +
  
  "aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=1.5[a1];" +
  
  "[0:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=3.5[a2];" +
  
  " [a1][a2]amerge,pan=stereo|c0code>

  how ever this is the problem I have faced

  It does add audio of 85 sec to video from 2:30 to 3:55 , but it starts from 0:00 of audio to 1:25 part of audio,

  What I want is to have a 22 sec clip of audio from (1 min 25 sec to 1 min 47 sec part of audio file) and loop it for 85 seconds and add that to video file

  how can that be done

• ffmpeg / Audacity channel splitting differences

  14 mars 2018, par Adrian Chromenko

  So I’m working on a speech to text project using Python and Google Cloud Services (for phone calls). The mp3s I receive have one voice playing in the left speaker, the other voice in the right speaker.

  So during testing, I manually split the original mp3 file into two WAV files (one for each channel, converted to mono). I did this splitting through Audacity. The accuracy was about 80-90%, which was perfect for my purposes.

  However, once I tried to automate the splitting using ffmpeg (more specifically : ffmpeg -i input_filename.mp3 -map_channel 0.0.0 left.wav -map_channel 0.0.1 right.wav), the accuracy dropped drastically.

  I’ve been experimenting for about a week now but I can’t get the accuracy up. For what it’s worth, the audio files sound identical to the human ear. I found that when I increase the volume of the output files, the accuracy gets better, but never as good as when I did the splitting with Audacity.

  I guess what I’m trying to ask is, what does Audacity do differently ?

  here are the sox -n stat results for each file :

  **Split with ffmpeg( 20-30% accuracy) : **

  Samples read:           1690560
  
  Length (seconds):    211.320000
  
  Scaled by:         2147483647.0
  
  Maximum amplitude:     0.433350
  
  Minimum amplitude:    -0.475739
  
  Midline amplitude:    -0.021194
  
  Mean    norm:          0.014808
  
  Mean    amplitude:    -0.000037
  
  RMS     amplitude:     0.028947
  
  Maximum delta:         0.333557
  
  Minimum delta:         0.000000
  
  Mean    delta:         0.009001
  
  RMS     delta:         0.017949
  
  Rough   frequency:          789
  
  Volume adjustment:        2.102

  Split with Audacity : (80-90% accuracy)

  Samples read:           1689984
  
  Length (seconds):    211.248000
  
  Scaled by:         2147483647.0
  
  Maximum amplitude:     0.217194
  
  Minimum amplitude:    -0.238373
  
  Midline amplitude:    -0.010590
  
  Mean    norm:          0.007423
  
  Mean    amplitude:    -0.000018
  
  RMS     amplitude:     0.014510
  
  Maximum delta:         0.167175
  
  Minimum delta:         0.000000
  
  Mean    delta:         0.004515
  
  RMS     delta:         0.008998
  
  Rough   frequency:          789
  
  Volume adjustment:        4.195

  original mp3 :

  Samples read:           3379968
  
  Length (seconds):    211.248000
  
  Scaled by:         2147483647.0
  
  Maximum amplitude:     1.000000
  
  Minimum amplitude:    -1.000000
  
  Midline amplitude:    -0.000000
  
  Mean    norm:          0.014124
  
  Mean    amplitude:    -0.000030
  
  RMS     amplitude:     0.047924
  
  Maximum delta:         1.015332
  
  Minimum delta:         0.000000
  
  Mean    delta:         0.027046
  
  RMS     delta:         0.067775
  
  Rough   frequency:         1800
  
  Volume adjustment:        1.000

  One thing that stands out to me is that the duration isn’t the same. Also the amplitudes. Can I instruct ffmpeg what the duration is when it is doing the splitting ? And can I change all the amplitudes to match the audacity file ? I’m not sure what to do to get to the 80% accuracy rate, but increasing volume seems to be the most promising solution so far.

  Any help would be greatly appreciated. I don’t have to use ffmpeg, but it seems like my only option, as Audacity isn’t scriptable.