Convert audio files to mp3 using ffmpeg

Question

I need to convert audio files to mp3 using ffmpeg   When I write the command as ffmpeg -i audio ogg -acodec mp3 newfile mp3  I get the error   FFmpeg version 0 5 2  Copyright  c  2000-2009 Fabrice Bellard  et al    configuration     libavutil     49 15  0   49 15  0   libavcodec    52 20  1   52 20  1   libavformat   52 31  0   52 31  0   libavdevice   52  1  0   52  1  0   built on Jun 24 2010 14 56 20  gcc  4 4 1 Input  0  mp3  from  ZHRE mp3     Duration  00 04 12 52  start  0 000000  bitrate  208 kb s     Stream  0 0  Audio  mp3  44100 Hz  stereo  s16  256 kb s Output  0  mp3  to  audio mp3       Stream  0 0  Audio  0x0000  44100 Hz  stereo  s16  64 kb s Stream mapping    Stream  0 0 - gt   0 0 Unsupported codec for output stream  0 0   I also ran this command    ffmpeg -formats   grep mp3   and got this in response   FFmpeg version 0 5 2  Copyright  c  2000-2009 Fabrice Bellard  et al    configuration     libavutil     49 15  0   49 15  0   libavcodec    52 20  1   52 20  1   libavformat   52 31  0   52 31  0   libavdevice   52  1  0   52  1  0   built on Jun 24 2010 14 56 20  gcc  4 4 1  DE mp3             MPEG audio layer 3  D A    mp3             MP3  MPEG audio layer 3   D A    mp3adu          ADU  Application Data Unit  MP3  MPEG audio layer 3   D A    mp3on4          MP3onMP4  text2movsub remove extra noise mov2textsub mp3decomp mp3comp mjpegadump imxdump h264 mp4toannexb dump extra   I guess that the mp3 codec isn t installed  Am I on the right track here

User · Answer

Try FFmpeg Static Build Link

Documentation: https://www.johnvansickle.com/ffmpeg/

Host the static build on your server in same directory

$ffmpeg = dirname(__FILE__).'/ffmpeg';

$command = $ffmpeg.'ffmpeg -i audio.ogg -acodec libmp3lame audio.mp3';

shell_exec($command);

User · Answer

High quality for Mac OS works perfectly   ffmpeg -i input wma -q a 0 output mp3

User · Answer

As described here input and output extension will detected by ffmpeg so there is no need to worry about the formats  simply run this command   ffmpeg -i inputFile ogg outputFile mp3

User · Answer

I had to purge my ffmpeg and then install another one from a ppa   sudo apt-get purge ffmpeg sudo apt-add-repository -y ppa jon-severinsson ffmpeg  sudo apt-get update  sudo apt-get install ffmpeg   Then convert          ffmpeg -i audio ogg -f mp3 newfile mp3

User · Answer

1  wav to mp3  ffmpeg -i audio wav -acodec libmp3lame audio mp3   2  ogg to mp3  ffmpeg -i audio ogg -acodec libmp3lame audio mp3   3  ac3 to mp3  ffmpeg -i audio ac3 -acodec libmp3lame audio mp3   4  aac to mp3  ffmpeg -i audio aac -acodec libmp3lame audio mp3

User · Answer

If you have a folder and sub-folder full of wav s you want to convert  put below command in a file  save it in a  bat file in the root of the folder where you wan to convert  and then run the bat file  for  R   g in    wav  do start  b  wait     C  ffmpeg-4 0 1-win64-static bin ffmpeg  -threads 16 -i    g  -acodec libmp3lame     dpng mp3   amp  amp  del    g

User · Answer

https   trac ffmpeg org wiki Encode MP3  VBR Encoding   ffmpeg -vn -ar 44100 -ac 2 -q a 1 -codec a libmp3lame output mp3

User · Answer

For batch processing with files in folder aiming for 190 VBR and file extension    mp3 instead of  ac3 mp3 you can use the following code  Change  ac3 to whatever the source audio format is   ffmpeg mp3 settings  for f in   ac3   do ffmpeg -i   f  -acodec libmp3lame -q a 2    f     mp3   done

User · Answer

Never mind    I am converting my audio files to mp2 by using the command   ffmpeg -i input wav -f mp2 output mp3   This command works perfectly   I know that this actually converts the files to mp2 format  but then the resulting file sizes are the same

User · Answer

For batch processing files in folder   for i in   wav  do ffmpeg -i   i  -f mp3    i   mp3   done   This script converts all  wav  files in folder to mp3 files and adds mp3 extension  ffmpeg have to be installed   See other answers

User · Answer

You could use this command   ffmpeg -i input wav -vn -ar 44100 -ac 2 -b a 192k output mp3   Explanation of the used arguments in this example    -i - input file -vn - Disable video  to make sure no video  including album cover image  is included if the source would be a video file -ar - Set the audio sampling frequency  For output streams it is set by default to the frequency of the corresponding input stream  For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options  -ac -     Set the number of audio channels  For output streams it is set by default to the number of input audio channels  For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options  So used here to make sure it is stereo  2 channels  -b a -  Converts the audio bitrate to be exact 192kbit per second

[audio] Convert audio files to mp3 using ffmpeg

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