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?

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
  

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

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

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)

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..

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

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

VBR Encoding:

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

High quality for Mac OS works perfectly!

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

No one seems to use find, which let you do everything on one line. Based on this answer and this post

find . -type f -iname "*.webm" -exec bash -c 'FILE="$1"; ffmpeg -i "${FILE}" -vn -ab 128k "${FILE%.webm}.mp3";' _ '{}' \;

For for eg podcasts 128k is enaugh for me. You can adjust that argument beside some others:

• -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

• Also, this notes the difference for the ffmpeg bitrate arguments

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);

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"

I will explain how to convert webm to mp3 for macs, I guess for linux it also works.

1. Install ffmpeg - brew install ffmpeg (mac) or sudo apt install ffmpeg (linux)
2. Create shell script - Open text editor put the following code inside:

#!/bin/bash

echo webm to mp3 converter! Work begins! 
for FILE in *.webm; do     
	echo -e "Processing file '$FILE'";
	ffmpeg -i "${FILE}" -vn -ab 128k -ar 44100 -y "${FILE%.webm}.mp3";
done;

this code will look all files with .webm extension in current directory.

3. Save this file without extension (for example "my-converter")
4. Navigate to created file via terminal
5. Make the file executabe by typing command: chmod 700 my-converter, now in the same directory the unix executable file (.sh) will be created.
6. Execute the file from terminal by typing: ./my-converter and the process begins, you will see the progress in the terminal window.

Done.

Using the previous answers, here is an alias for this by adding the following into .bashrc/.zshrc:

alias convert-aac="cd ~/Downloads && aac-to-mp3"

# Convert all .aac files into .mp3 files in the current folder, don't convert if a mp3 file already exists
aac-to-mp3(){
    find . -type f -iname "*.aac" -exec \
        bash -c 'file="$1"; ffmpeg -n -i "$file" -acodec libmp3lame "${file%.aac}.mp3";' _ '{}' \;
}

Usage: convert-aac (in shell)

>Thanks to https://stackoverflow.com/a/70339561/2391795 and https://stackoverflow.com/a/12952172/2391795 and https://unix.stackexchange.com/a/683488/60329