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Comment by mwint | Hacker News Reader

Completely idle curiosity: What kind of artifacts? I don’t spend much time on AV stuff, but I always find the details fascinating.

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One obvious one is the audio quality of Twitter videos, or lack thereof. That one immediately told me they were using ffmpeg's AAC encoder (it's fairly obvious on piano music). Some of ffmpeg's built in codecs are great and some aren't; AAC is in the latter category. I've done some ABX tests with it and some transcoding tests and it's just not great; even at 320kbps two or three transcode cycles result in clearly audible artifacts, especially in high frequency sounds like cymbals (you get a sort of warbly characteristic). MP3 actually has better performance in ffmpeg (via LAME) than AAC at the same bitrate, using the built-in encoder.

If you've ever heard a Twitch streamer, especially people playing 8-bit games or chiptune-ish music, and wondered why the audio sounded a bit dodgy, it's ffmpeg. AAC at 160kbps (the streaming standard) is supposed to sound better than that. OBS will use FDK if available, and on macOS and Windows it'll use CoreAudio if installed, but you need to install an old MSI pulled from an old version of iTunes to get that on Windows... so the vast majority of streamers on Windows are stuck with ffmpeg-AAC.

If you're using ffmpeg on Linux, install libfdk and use libfdk_aac; if you're on macOS, use aac_at instead. Those are much better encoders and ffmpeg can use them as external libraries.

Note that this isn't a criticism of ffmpeg; lack of quality in some codecs is a direct result of lack of funding (though I do fault them for, at one time, claiming the built in AAC encoder was good enough to be made the default and competitive with FDK; it wasn't and still isn't, but I think they reverted that claim at some point).

(source: I help run a couple anime DJ restreaming-based events and fixing the AAC issue is a big red warning item in my set-up guide; I once got a double transcode feed where both steps were using ffmpeg-AAC and I could instantly tell something was wrong)

For a few years, the default aac coder has been fast. But the twoloop coder is better but slower, and was made default in May 2021. Add `-aac_coder twoloop` (in older builds) and recheck.

I should do a comparison with the more recent one then; it's been a while since I first ran into this. Thanks for the pointer!

> if you're on macOS, use aac_at instead. Those are much better encoders and ffmpeg can use them as external libraries.

Hey, do you happen to know if aac_at is encoding at the slowest speed? (Most compression efficiency but longest encoding time). I can't figure out for sure, so I keep bouncing audio through afconvert since it lets me set `-q 127`, and it's a pain, I'd rather do one step.

There are two quality parameters and I'm not sure which one is `-q` in afconvert. One of them is the VBR quality, `kAudioCodecPropertySoundQualityForVBR`.

ffmpeg maps it like this:

q = 127 - q * 9;

So `-q:a 0` should do the same thing as passing 127 to AT.

There is another property though; an actual control of codec effort (`kAudioConverterCodecQuality`). That one should be accessible as `aac_at_quality` and maps 0 to 96, so there's no way to go up to the max of 127... But Apple defines "High" as 96, so I get the feeling there won't be a big difference between that and max, especially for an audio codec.

Thank you, this is the best explanation I've gotten in years!

> That one should be accessible as `aac_at_quality` and maps 0 to 96, so there's no way to go up to the max of 127.

Is this a bug I should report to ffmpeg?

If there is a nontrivial file size or quality difference between those two settings, sure. If not, it's probably not worth it :)

Would you simplify your ffmpeg advice for the less-programmerly among us?

I use ffmmpeg all the time on my M1 Max and if there are parameters that I should always be using, I would like to know :)

> Windows it'll use CoreAudio if installed, but you need to install an old MSI pulled from an old version of iTunes to get that on Windows... so the vast majority of streamers on Windows are stuck with ffmpeg-AAC.

I think windows might have a built in AAC encoder in media foundation? Wonder if that's true and if it's any good.

IIRC there was some attempt at supporting it but it was broken or not very good? I forget...

Why in the world are you still fooling with AAC? Everybody supports Opus now.

Live streaming is stuck in the dark ages of RTMP with H.264 and AAC. It's basically hardcoded everywhere, there is barely any support for other codecs or protocols. You want to send streams to Twitch or YouTube, you're sending H.264 and AAC.

OBS doesn't even let you pick streaming codecs. It just assumes H.264 and AAC. You can get it to stream in other codecs by abusing the "record" function with an output to stream instead of filename, but it's a hack...

not even ffmpeg can bring an encoding back from the dregs

When transcoding some streams, if you use ffmpeg's defaults, sometimes the way it handles PTS, DTS and keyframes isn't ideal, and you'll get weird video pauses, blanks, and strange but distinct visual artifacts while the audio plays just fine.

Having spent a lot of time using ffmpeg for transcoding, it's one of those things I recognized because I'd often forget to set the correct settings and got similar results, as well.

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