Ffmpeg FLAC to AAC encoding

aacaudioencodingffmpegflac

I wanted to make sure that I'm not shooting myself in the foot and ask about this.
So currently I have an audio file in a .mkv container that looks like this

Audio: FLAC (framed) 48000Hz stereo 1536kbps [A: Audio (flac, 48000 Hz, stereo, s16) [default]]

ID : 2
Format : FLAC
Format/Info : Free Lossless Audio Codec
Codec ID : A_FLAC
Duration : 39mn 50s
Bit rate mode : Variable
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Title : Audio
Writing library : libFLAC 1.2.1 (UTC 2007-09-17)

And I would like to convert it to AAC, preferably without losing the quality too much.
I did find a similar question asked on this forum.
But the method that was described there is about 3 years old.

In short:

"compile your own ffmpeg for fdk_aac
follow the AAC encoding guide and use fdk_aac's -vbr option – with a setting of 3"

Resulting in:

ffmpeg -i input.flac -c:a libfdk_aac -vbr 3 output.m4a".

By evilsoup. – taken from here.

So I was wondering if this will work for my situation, or if it won't work for me.
If that's all it takes then no need to answer, and thanks!

Best Answer

FFmpeg supports two AAC-LC encoders:

Native FFmpeg AAC encoder (aac)
Fraunhofer FDK AAC (libfdk_aac)

FFmpeg supports one HE-AAC encoder:

Fraunhofer FDK AAC (libfdk_aac)

libfdk_aac

libfdk_aac is still the best AAC-LC encoder supported by FFmpeg, and is the only HE-AAC encoder supported by FFmpeg. However, its license is problematic when it comes to distribution, so you'll need to compile ffmpeg to be able to use it.

Native FFmpeg AAC encoder

This encoder has seen many improvements since the answer you refer to. It is no longer considered experimental, but it is not yet as good as libfdk_aac. However, if you give it enough bits it should be acceptable. The advantage of this encoder is that it is always available and comes with FFmpeg.

Pipe

Not always practical, but you can pipe to your favorite, standalone encoder:

ffmpeg -i input -f wav - | fdkaac -I -m 5 - -o output.m4a

Also see

FFmpeg Wiki: AAC

Related Solutions

FFmpeg settings for YouTube and Facebook video uploads

Don't consider the guidelines as strict requirements. General recommendation is to provide the highest quality that is practical for you to upload. It's that simple; whatever you upload is going to be re-encoded anyway and YouTube will almost always accept whatever you give them. That means you either upload the original content, or if the original is too big you can re-encode it using a high quality. Example using ffmpeg:

ffmpeg -i input -c:v libx264 -crf 18 -preset slow -c:a copy output.mkv

See FFmpeg Wiki: H.264 for more details–specifically the -crf and -preset options. Notice that I simply copied the audio, but you may choose to re-encode it if the source contains uncompressed audio.

Your player may not be able to play the output for various reasons, but YouTube certainly will.

Recommended settings to convert Opus to AAC

More generally, I would recommend to have a look at the FFmpeg Wiki page on AAC encoding. Here you will find tips on how to achieve the best quality, which involves:

using the libfdk_aac encoder instead of the built-in one (the encoder can make a huge difference in quality!)
using VBR — there is no need to constrain the rate or waste bits
turning off the high pass filter, as by default some high frequencies are lost during conversion

Thus, the following would work:

ffmpeg -i input.opus -c:a libfdk_aac -vbr 5 -cutoff 18000 output.m4a

This should be absolutely transparent to anyone's ears. You may even use -vbr 4 and check whether you can detect the difference. I guess you won't.

More specifically, you ask:

And finally, youtube-dl also can download an AAC file of 128 kbs. (…) But would transcoding the opus to AAC (…) maintain that quality difference, or would the transcoding (…) induce artefacts so that the audio quality (…) would actually be worse?

Great question that I don't know how to answer without conducting a formal listening test.

However, if you assume that stereo AAC at 128 kbps is at least good quality (but not excellent), you may forgo any conversion and use that directly if it sounds good to you. I think I would listen to the original Opus and AAC samples that you directly downloaded and try to do a blind test — can you reliably detect the difference? If no, then just use AAC and save yourself the conversion.

PS: I listened to a few music clips from YouTube with both the Opus and AAC variants and I personally liked Opus better — it seemed to have a bit more high-frequency spatiality to it. But don't consider that a scientifically sound conclusion!

Finally, there is a thread discussing a similar issue, and here, one user thinks that there is more aggressive psycho-acoustic shaping in VBR encoding, which may enhance already present artifacts when encoding a lossy file again. He concludes that ABR at a high bitrate may be better suited for such tasks.