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 -> #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?

score 454 · Answer 1 · edited Apr 22 '23 at 10:15

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 192k - Converts the audio bit-rate to be exact 192 KB/s (192 kibibit per second).

But maybe use -q:a 2 instead, which allows the encoder to pick from 170 to 210 KB/s quality-range (average 192 KB/s). But -q format may not be compatible with some old player-hardware.

Note to see docs about bit-rate argument's differences. Because maybe that option is the most important one, as it decides the "quality" versus "output size" versus "old mp3-player compatibility".

Where:

-b:a is for CBR (constant-bit-rate), which should be compatible with most old players, but may take more file-size.

-q:a or -qscale:a alias, is for VBR (variable-bit-rate).

--abr is for ABR (adaptive-bit-rate), which is a combo of CBR and VBR modes, but --abr argument needs -b to be passed as well (because ffmpeg does not take any parameters after --abr, unlike lame --abr executable).

@apanloco for me, changing -b to -q absolutely butchers the sound. Using no options at all, or using the options presented in the answer, sound virtually the same as the source .wav. — michael_teter, Aug 05 '19 at 00:00
Cool to batch `for f in *.wma; do ffmpeg -i "$f" -vn -ar 44100 -ac 2 -b:a 192k "${f%.*}.mp3"; done` — Ax_, Apr 13 '22 at 03:20
@Ax_ It's important to note that your solution will only work on non-Windows installs. To do this from the Windows command line, you can use this: `for %f in (*.wma) do ffmpeg.exe -i "%f" -vn -ar 44100 -ac 2 -b:a 192k "%f.mp3"` This will create files for every wma file in the current folder, with the original name and ".mp3" appended to it after the ".wma". E.g. `input.wma -> input.wma.mp3` — sosaisapunk, Jul 08 '22 at 14:57

score 176 · Answer 2 · edited Apr 17 '23 at 11:49

176

wav to mp3

ffmpeg -i audio.wav -acodec libmp3lame audio.mp3

ogg to mp3

ffmpeg -i audio.ogg -acodec libmp3lame audio.mp3

ac3 to mp3

ffmpeg -i audio.ac3 -acodec libmp3lame audio.mp3

aac to mp3

ffmpeg -i audio.aac -acodec libmp3lame audio.mp3

edited Apr 17 '23 at 11:49

General Grievance

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answered May 18 '16 at 11:39

Ijas Ahamed N

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How do you specify the mp3 bitrate? – Suragch Dec 14 '16 at 12:14
8

Add `-b:a 128k` for 128 kbps. – Gyan Dec 18 '16 at 11:35
11

Or even shorter without `acodec`: ffmpeg -i audio.aac audio.mp3 – Timo Jul 11 '18 at 06:57
how to convert from .mp3 to .ogg ? – Cristea Victor Jun 01 '20 at 15:38
I use libmp3lame but why its codec on VLC says mp2 (mpeg audio layer 1/2)??? – Fthr Feb 04 '21 at 03:41
@Timo The -acodec libmp3lame is necessary for me, when transferring from aac to mp3. – Millar248 Nov 05 '21 at 13:29
Is there any solution for audio/mp4 to mp3? – awais Apr 13 '22 at 12:27

planb · Answer 3 · 2017-01-04T12:17:19.190

36

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

edited Jan 04 '17 at 12:17

answered Jan 04 '17 at 11:58

planb

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MKK · Answer 4 · 2016-12-18T15:39:57.153

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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)

edited Dec 18 '16 at 15:39

answered Dec 18 '16 at 10:28

MKK

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With `-f mp2` MP2 is generated, not MP3. Change it to `-f mp3` – Gyan Dec 18 '16 at 11:35
10

The above command creates files that are named `wav.mp3`. To get files with the correct file extension, change the command to: `for i in *.wav; do ffmpeg -i "$i" -f mp3 "${i%.*}.mp3"; done`, i.e. add `.*` after `i%`. – Suzana Feb 25 '18 at 14:16

score 26 · Answer 5 · edited Nov 29 '19 at 14:25

26

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

edited Nov 29 '19 at 14:25

Steak Overflow

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answered Oct 18 '19 at 01:33

Kasra

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score 18 · Accepted Answer · answered Jul 24 '10 at 19:06

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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..

answered Jul 24 '10 at 19:06

Hrishikesh Choudhari

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Generates a wave file – LtWorf Oct 06 '14 at 17:58
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This does not result in a mp3 file – m13r Oct 17 '14 at 20:53
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While the sizes are the same, **these mp2 files are not playable in a web browser**, as they are not mp3. Not even with a flash player. The only benefit they have is that they require less cpu to encode, but the drawbacks are not worth it. – Daniel F Apr 15 '18 at 21:29
8

This should not be the accepted answer, as it does not answer the question correctly. – reinierpost Jun 16 '18 at 21:28
To reduce the size use this: https://superuser.com/a/553049/1623401 – Kamal Hossain Jan 03 '22 at 18:04
It is a bad idea to suffix mp2 audios as .mp3 because it could be badly detected as mp3. – FFmpegEnthusiast Apr 12 '23 at 15:29

score 13 · Answer 7 · answered Jan 09 '15 at 11:54

13

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

answered Jan 09 '15 at 11:54

Komu

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this is the short way, but if you want to specify the quality kb per seconds, use -ab 192k – wrivas Aug 29 '15 at 13:57

Pablo Bianchi · Answer 8 · 2023-03-06T03:56:05.127

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 e.g. podcasts, 128k is enough 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

score 4 · Answer 9 · edited Apr 15 '23 at 14:34

4

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

VBR Encoding:

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

edited Apr 15 '23 at 14:34

FFmpegEnthusiast

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answered Jun 28 '18 at 13:20

Guillaume M.

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You forgot the input. – llogan Mar 29 '19 at 17:32

score 3 · Answer 10 · answered Dec 22 '18 at 09:00

3

High quality for Mac OS works perfectly!

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

answered Dec 22 '18 at 09:00

Ivan Sanchez

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M22 · Answer 11 · 2021-04-13T21:38:44.557

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

Install ffmpeg - brew install ffmpeg (mac) or sudo apt install ffmpeg (linux)
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.

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

Done.

Code Book · Answer 12 · 2020-09-14T05:19:19.893

1

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);

edited Sep 14 '20 at 05:19

answered Jan 26 '18 at 14:17

Code Book

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score 1 · Answer 13 · answered Jul 11 '18 at 19:52

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"

score 0 · Answer 14 · answered Dec 22 '21 at 13:27

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

score -1 · Answer 15 · answered Apr 16 '23 at 03:18

-1

for file in *.wma; do ffmpeg -i "${file}"  -acodec libmp3lame -ab 192k "${file/.wma/.mp3}"; done

answered Apr 16 '23 at 03:18

KingLiu

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Remember that Stack Overflow isn't just intended to solve the immediate problem, but also to help future readers find solutions to similar problems, which requires understanding the underlying code. This is especially important for members of our community who are beginners, and not familiar with the syntax. Given that, **can you [edit] your answer to include an explanation of what you're doing** and why you believe it is the best approach? – Jeremy Caney Apr 17 '23 at 01:59

Convert audio files to mp3 using ffmpeg

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