Comment by daneel_w
2 years ago
A couple more:
afconvert(1) - an audio file format converter, which includes Apple's superior AAC codec from the Core Audio framework
diskutil(8) - tons of tools for fixed and removable storage
Examples:
afconvert in.wav -o out.m4a -q 127 -s 2 -b 160000 -f m4af -d 'aac '
mb=300; diskutil eraseVolume APFS myramdisk `hdiutil attach -nomount ram://$((mb*2048))`
Mostly I use XLD (https://tmkk.undo.jp/xld/index_e.html) for audio conversion (as I'm mostly converting from .BIN + .CUE to "iTunes Plus" AAC for uploading to iTunes Match); but my understanding is that under the covers it's mostly just using afconvert (or whatever the system-framework equivalent of it is.)
So if your needs are just "one audio file in, one audio file out, and let me tell you exactly what it should look like", then afconvert is probably what you want.
XLD is great. The best thing about it is the heavy parallelization if you have many cores to throw at the problem. You can convert mountains of albums very fast.
Love those type of OG Mac indie software.
Yes, iTunes also uses the same Core Audio framework. In fact, even iTunes on Windows uses the same code base through the Quicktime-for-Windows libraries it runs off of.
I prefer to encode with afconvert on the command line because it gives me a few more options for tweaking things, that I don't have access to in iTunes (or "Music" as it's called these days). Additionally, I use a simple shell script that handles all of it when for example ripping a whole CD.
+1 to XLD. I have been using it for years, it's a wonderful piece of software.
> afconvert
Oh wow! A while back I ripped some concert audio from Youtube, but it was too large for me to sync using my `iTunes Match`. I've been too lazy to figure out how to downsample it juuuuust enough. But it looks like this works right out of the box
"superior" ... It looks silly to include such a judging adjective in the description of a comman line utility. I don't need "subtle" Apple ads on the command line. And aac is about the last format I would choose, encoding my audio in.
To be very clear, the man page is claiming that Apple's implementation of the AAC codec [the encoder specifically] is superior to other implementations of AAC encoding; not that AAC is superior to other audio codecs.
If you're wondering how one implementation of an audio codec could be superior to others — mostly, it's because any lossy audio codec has an encoding phase called "psychoacoustic compression", where each implementation of the codec is free to do whatever it likes to "simplify" the waveform in some way (most easily pictured: by taking a Discrete Cosine Transform of the waveform, and then quantizing / convoluting some parts of the frequency space. Like what JPEG does to discard information from the chroma channels.)
IIRC, rather than blunt-force quantization, Apple's AAC encoding does clever things (akin to the instrument separation done to audio in Melodyne) to split the waveform into "features", and then discards information in such a way that the features' separability is maintained (i.e. it doesn't become harder to "pick out" any given "sound" from the audio.)
Correct. Though AAC (when produced with a proper encoder) surpasses all other lossy formats with the exception of Opus.
I think those are the commenter's descriptions, not Apple's man page synopses. My system has
Its quality is technically superior as evident from the product's PSNR and other factors. There are no "ads" here, I'm not a brand warrior, and nobody cares what formats are the last ones you'd encode your audio in.
There's also afplay - Audio File Play (plays an audio file to the default audio output). Can play all audio formats supported by CoreAudio.
How does that compare with ffmpeg? The arguments seem about the same.
afcovert uses the superior inbuilt AAC converter. FFmpeg can do this as well with the right arguments but you have to dig them up and the quality is capped to a lower value than you can get with afconvert.
Isn't -c:a aac_at the same? And then designate bitrate?
There's no reason to use AAC when we have both OPUS and FLAC.
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ffmpeg offers a few different encoders for many of its supported audio/video formats, and the result depends on what encoder you tell ffmpeg to use. It does not have support for using Core Audio on macOS but it does offer FDK AAC, which is one of the better AAC encoders available today - though not nearly as good as the one available in Core Audio.
Do you have any sources/informations on an objective comparison of encoders quality ?
Way back then (iPod era), I made an exposé for a science exam (small paper), proving that AAC was indeed better than MP3 both subjectively and objectively at most bitrates. This is how I got introduced to Fourier transforms and the likes; it was very interesting to see that you can literally "see" the difference in quality on the encoded waveform output.
But I just used the default encoders on the Mac and I didn't think about going in deep into encoder comparisons at the time. Does it matter that much? From what I know, it just about properly programming a math spec, with some tricks but I wonder if that makes that much of a difference.
In any case, above 256kbps it takes a very skilled listener to correctly identify encoded music. Apple has some useful tools for that, particularly AU Lab that allows you A/B testing of tracks with on-the-fly encoding. https://www.apple.com/apple-music/apple-digital-masters/
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It does. -c:a aac_at uses Core Audio (_at means AudioToolbox).
> afconvert
See also XACT[1]. It's a macOS GUI, but it's nerd-friendly.
[1] http://xact.scottcbrown.org