[audio] Convert audio files to mp3 using ffmpeg

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 -> #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?

High quality for Mac OS works perfectly!

ffmpeg -i input.wma -q:a 0 output.mp3

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);

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)

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

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

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

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

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

https://trac.ffmpeg.org/wiki/Encode/MP3

VBR Encoding:

ffmpeg -vn -ar 44100 -ac 2 -q:a 1 -codec:a libmp3lame output.mp3

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" && del "%%g"