Converting multiple flac files to mp3 using the CLI

[rmcellig]

This is new territory for me. I previously relied on GUI apps to batch convert files from Flac to mp3 and wav and aiff to mp3.

How would I do this using ffmpeg?

I have several folders with aiff, flac and wav files that I want to convert to 192 bit mp3 format. I have a feeling using ffmpeg would be much faster and easier to use than a gui solution. The other reason is that I am determined to use the CLI for most of my tasks instead of always relying on a GUI solution. So far, I am really discovering the power of the CLI and I am really enjoying it (never thought I would say that in a million years .

Thanks!!

[varunendra]

Thread moved to Multimedia & Video.
..and bumped

[andrew.46]

Probably the easiest way is to use a for loop, but it depends what sort of directory structure you have in place. Are the files all lumped together or separated by type?

[fdrake]

This is new territory for me. I previously relied on GUI apps to batch convert files from Flac to mp3 and wav and aiff to mp3.

How would I do this using ffmpeg?

I have several folders with aiff, flac and wav files that I want to convert to 192 bit mp3 format. I have a feeling using ffmpeg would be much faster and easier to use than a gui solution. The other reason is that I am determined to use the CLI for most of my tasks instead of always relying on a GUI solution. So far, I am really discovering the power of the CLI and I am really enjoying it (never thought I would say that in a million years .

Thanks!!

mp3 format losses quality when formating to it. Can you use ogg instead?

mkdir /my_ogg_dir
for i in /my_Flac_dir/*.flac
do
ffmpeg -i $i -strict experimental -acodec vorbis -aq 100 /my_ogg_dir/$i.ogg
done

see also : http://ubuntuforums.org/showthread.php?t=1626645

[evilsoup]

Ogg Vorbis is a lossy codec, just like MP3. Well, Vorbis is better than MP3 at a given bit rate - it's on par with AAC-LC - but that's only applicable to very low bit rates; for most people, using consumer audio equipment, most of the time all modern lossy audio encoders produce results indistinguishable from the original CD audio at around 128 kbit/s.

Vorbis is nowhere near as well supported as MP3 on hardware players. Also, the ffmpeg vorbis encoder is sub-par; you should use libvorbis instead.

ffmpeg -i input.flac -c:a libvorbis -q:a 5 output.ogg

To actually answer the OP question:

##  Encode a single file as an MP3:
ffmpeg -i input.flac -c:a libmp3lame -q:a 4 output.mp3

##  Encode a directory of FLAC files to MP3:
for f in *.flac; do ffmpeg "$f" -c:a libmp3lame -q:a 4 "${f/%flac/mp3}"; done

##  Create MP3s recursively (make sure you copy all the quotes):
find . -type f -name "*.flac" -exec bash -c 'ffmpeg -i "$0" -c:a libmp3lame -q:a 4 "${0/%flac/mp3}"' '{}' \;

See this blog post for some basic information about encoding different formats; see this page on the ffmpeg wiki for a bit more information on MP3 encoding (and this HydrogenAudio page for some in-depth information).

[shantiq]

hi rm this will do it if you cd to the folder where all your folder files are say they are all in Music do

cd Music

nullglob means it does not stop if one of the formats listed is absent nocaseglob means it cares not if .mp3 or .MP3

for i in */
do
cd "$i"
for f in nullglob nocaseglob *.{flac,aiff,wav} ; do ffmpeg -i "$f" -b:a 192k "${f%.*}.mp3" ; done
cd ..
done

if you want to remove original files be careful here you can add [test it first with lines above]

for i in */
do
cd "$i"
for f in nullglob nocaseglob *.{flac,aiff,wav} ; do ffmpeg -i "$f" -b:a 192k "${f%.*}.mp3" ; done \
&& rm *.{flac,aiff,wav}
cd ..
done

PS love your concise line Esoup

find . -type f -name "*.flac" -exec bash -c 'ffmpeg -i "$0" -c:a libmp3lame -q:a 4 "${0/%flac/mp3}"' '{}' \;

how do you add wav and aiff to it? [i tried to add those and messed up with find syntax]

[evilsoup]

I got that find line originally from this superuser.com answer, so if you have an account there, give that guy an upvote.

I would use regular expressions for finding multiple file extensions, something like:

find . -type f -regextype posix-extended -regex ".*flac|.*wav|.*aiff" -exec bash -c 'ffmpeg -i "$0" -c:a libmp3lame -q:a 4 "${0%.*}.mp3"' '{}' \;

##  or (slightly more concise, easier to add more file extensions):
find . -type f -regextype posix-extended -regex ".*\.(flac|wav|aiff)" -exec bash -c 'ffmpeg -i "$0" -c:a libmp3lame -q:a 4 "${0%.*}.mp3"' '{}' \;

You may have to escape the '|'s, unfortunately my linux computer has suffered physical damage & I'm stuck on Windows 8 til Tuesday, so I'm unable to test if this exact syntax works. Check out the find manpage, I guess. You can also use something like:

find . -type f -name "*.flac" -o -name "*.wav" -o -name "*.aiff" -exec bash -c 'ffmpeg -i "$0" -c:a libmp3lame -q:a 4 "${0/%flac/mp3}"' '{}' \;

...but I prefer the regex way, since it's more concise.

You can also use '-iregex' instead of '-regex' (or '-iname' instead of '-name') to get case-insensitive matches - so it would match both 'whatever.wav' and 'whatever.WAV'

[shantiq]

thanx man really good ; even goes into folders within folders [truly recursive] way more elegant than my paltry earlier proposition
just noticed -q:a 4 does not really hit 192k the asker requested so it seems more accurate to use -b:a 192k instead what do you think ? and then the iregex is a peach addition , maybe can also trim -c:a libmp3lame to make it even leaner [all recent versions of ffmpeg ok that way]

this is very elegant .... shall be using from now on and ditching the old method

find . -type f -regextype posix-extended -iregex ".*\.(flac|wav|aiff)" -exec bash -c 'ffmpeg -i "$0" -b:a 192k "${0%.*}.mp3"' '{}' \;

[evilsoup]

Outside of specialized applications, VBR modes should be used for all video/audio encoding. '-q:a 4' will offer a very good level of audio transparency; using a lower value will get a higher bit rate (a value of 2 is equivalent - averaged over a large number of encoded files - to about 192 kbit/s), but I seriously doubt that anyone (outside of sound engineers with high-end equipment sitting in a recording studio & actively listening for artefacts... and even then...) will be able to tell the difference.

'-b:a 192k' will always get a bit rate of 192 kbit/s... but I think it would be pretty wasteful. Of course, hard drive space is pretty cheap, and it will subjectively sound as good as a VBR MP3 (again, to most people under most circumstances), so there's little real harm in doing so. I mostly recommend it because 'VBR=good' is a very good rule of thumb for delivery codecs, and is more important with video codecs, where hard drive space can still be an issue.

[evilsoup]

If yopu have a beefy multi-core machine, you may find some advantage in using gnu parallel Note: the parallel in the 12.04 repos is not gnu parallel (I don't know about 12.10 or above), get it from here.

FFmpeg is multithreaded, so this probably won't give you a massive boost unless you have a very beefy machine (>4 cores), but...

find . -type f -name "*.flac" | parallel ffmpeg -i {} -c:a libmp3lame -q:a 4 {.}.mp3

With GNU parallel, '{.}' is equivalent to bash '${f%.*}'. I don't think it's available in find -exec, hence the 'bash -c'.

Also, yes, the '-c:a libmp3lame' could be safely removed AFAICT.

EDIT: if any of the filenames contain newlines (should never happen...) you'll need to use this construction:

find . -type f -name "*.flac" -print0 | parallel -0 ffmpeg -i {} -c:a libmp3lame -q:a 4 {.}.mp3