How do I programmatically convert mp3 to an itunes-playable aac/m4a file?

爷,独闯天下 提交于 2019-11-30 15:31:18

FFmpeg provides AAC encoding facilities if you've compiled them in. If you are using Windows you can grab full binaries from here

ffmpeg -i source.mp3 -acodec libfaac -ab 128k dest.aac

I'm not sure how you would call this from ruby.

Also, be sure to set the bitrate appropriately.

There are only three free AAC encoders that I know of that are available through a commandline interface:

  1. FAAC (LPGL), which is honestly pretty bad (the quality is going to be significantly worse than LAME at the same bitrate). Its fine though if you're willing to go for higher bitrates (>>128kbps) and need AAC for compatibility, not quality reasons. The most common way to use FAAC is through ffmpeg, as libfaac.

  2. Nero AAC, the commandline encoder for which is available for free under Windows and Linux, but only for noncommercial use (and is correspondingly closed-source).

  3. ffmpeg's AAC encoder, which is still under development and while I believe it does technically work, it is not at all stable or good or even fast, since its still in the initial stages. Its also not available in trunk, as far as I know.

(Edit: Seems iTunes might have one too, I suspect its terms of use are similar to Nero's. AFAIK its quality is comparable.)

I realize I'm late to this party, but I'm questioning the premise of this question. Why do you even want to convert an MP3 to an "itunes playable" format? iTunes already handles MP3s natively.

It seems like you are doing an unnecessary conversion, and since you are converting from one lossy format to another, you are losing some quality in the process.

After installing the converting app on the linux/window machine you're running your Rails application on, use the "system()" command in Ruby to invoke the converting application on the system. system("command_here");

in ffmpeg 0.5 or later use ffmpeg -i source.mp3 target.m4a

for better results to transfer metadata and to override default bitrate ffmpeg applies

ffmpeg -i "input.mp3" -ab 256k -map_meta_data input.mp3:output.m4a output.m4a

best do not convert as ipod plays mp3 fine (I know there is such answer but my low standing does not allow voting)

I've had good luck using mplayer (which I believe uses ffmpeg...) and lame. To the point that I've wrapped it up in a script:

#!/bin/sh

TARGET=$1

BASE=`basename "${TARGET}"`
echo TARGET: "${TARGET}"
echo BASE:   "${BASE}" .m4a

# Warning! Race condition vulnerability here! Should use a mktemp
# variant or something...
mkfifo encode
mplayer -quiet -ao pcm -aofile encode "${TARGET}" &
lame --silent encode "${BASE}".mp3
rm encode

Sorry for the security issues, I banged this out on the train one day...

My mplayer and lame come from fink

Actually, syntax is ffmpeg -i input.mp3 -c:a aac -strict -2 -b:a 256k output.m4a; more correct if one is emulating "correct" bitrate. cf.:link for a compilation scheme. (rpmfusion package works fine too:

configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect

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