Both ALAC and FLAC are lossless audio formats and files will usually have more or less the same size when converted from one format to the other.
I use ffmpeg -i track.flac track.m4a to convert between these two formats but I notice that the resulting ALAC files are much smaller than the original ones. When using a converter software like the MediaHuman Audio Converter, the size of the ALACs will remain around the same size as the FLACs so I guess I'm missing some flags here that are causing ffmpeg to downsample the signal.
2 Answers
Ok, I was probably a little quick to ask here but for the sake of future reference here is the answer:
One should pass the flag -acodec alac to ffmpeg for a lossless conversion between FLAC and ALAC:
ffmpeg -i track.flac -acodec alac track.m4a
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9To explain what's happening here: .m4a is an Apple variant of the MP4 file format. FFmpeg and most other s/w will default to the AAC encoder when outputting to
mp4orm4a, hence the express-acodecoption is needed.– GyanCommented Jan 8, 2018 at 7:15 -
5Some FLAC files contain an album cover thumbnail. You can add
-vcodec copyto include those in your new ALAC files.– SeanCommented Nov 20, 2019 at 1:31
And for converting a whole directory...
Usage
pushd './Music/Some Album [flac]'
bash flac-to-alac.sh
flac-to-alac.sh:
#!/usr/bin/env bash
my_bin="$(dirname $0)/flac-to-alac-ffmpeg.sh"
find . -type f -name '*.flac' -exec "$my_bin" {} \;
flac-to-alac-ffmpeg.sh:
#!/usr/bin/env bash
set -e # fail if there's any error
set -u
my_file=$1
my_new="$(echo $(dirname "$my_file")/$(basename "$my_file" .flac).m4a)"
echo "$my_file"
ffmpeg -y -v 0 -i "$my_file" -acodec alac "$my_new"
# only gets here if the conversion didn't fail
#rm "$my_file"
Alternative:
I thought I could get this to work in a single command,
but it doesn't escape special characters, such as [.
It seemed so promising...
#!/usr/bin/env bash
set -e # exit immediately on error
set -u # error if a variable is misspelled
while read -r my_file; do
# ./foo/bar.flac => ./foo/bar.m4a
my_new="$(dirname "$my_file")/$(basename "$my_file" .flac).m4a"
ffmpeg -i "$my_file" -acodec alac "$my_new"
# safe because of set -e, but still do a test run
#rm "$my_file"
done <<< "$(find . -type f -name '*.flac')"
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8Here's a one-liner I use for conversion:
for i in *.flac; do echo $i; ffmpeg -i "$i" -y -v 0 -vcodec copy -acodec alac "${i%.flac}".m4a && rm -f "$i"; doneCommented Mar 13, 2020 at 7:52 -
@PaulLindner That looks like a perfect option for single directories, no recursion.– coolaj86Commented Mar 13, 2020 at 17:29
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I did this (borrowing from @PaulLindner, whose solution didn't quite work on my system):
for i in *.flac; do echo $i; ffmpeg -i "$i" -y -acodec alac "${i%.flac}".m4a; doneCommented Feb 10, 2021 at 20:56 -
1audio files have the gnarliest filenames ever. I had
10ccand10CCconflicting on a big job recently. im using clementine and audiobrainz to sanitise it all.– TomachiCommented Oct 11, 2021 at 13:34 -
1I do NOT recommend deleting in the same command after encoding, do it manually to be sure, here is my version of the one-liner:
for i in *.flac; do ffmpeg -i "$i" -y -vn -c:a alac "${i%.flac}".m4a; doneCommented May 18, 2022 at 10:08
ffmpeggenerally needs the-acodecfor any destination to be sure you get the conversation done right. There are lots of front ends that useffmpegbut I've noticed many do not include ALAC as an output option.