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FFmpeg 9.1's new AAC encoder

hydrogenaudio.org

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Re: FFmpeg 9.1's new AAC encoder

#21

Post by palmotea »

cogman10 wrote: post #16

Man what a showcase for Opus this is. Don't get me wrong, this sort of thing is a valuable exercise and we are better off with better encoders for these older codecs. But look at the numbers for Opus on this benchmark. It simply blows all the AAC encoders out of the water even at 64 kbps.

> Man what a showcase for Opus this is.

I take it you mean this Opus (https://en.wikipedia.org/wiki/Opus_(audio_format)) not that Opus (https://en.wikipedia.org/wiki/Claude_(AI)).

I read almost all the way through your comment thinking there was a decent probability you were saying this new AAC encoder was written with Claude Opus.

Re: FFmpeg 9.1's new AAC encoder

#22

Post by repelsteeltje »

ndiddy wrote: post #17

Nice, I'm looking forward to seeing how this performs in practice. FFmpeg's previous AAC encoder produced poor quality output and often had irritating chirping artifacts, so I've always had to install Apple's Core Audio encoder on any computer I do video recording on to get decent sound. I've done A/B/X comparisons and found that a 320kbps MP3 sounds better than a 320kbps AAC encoded by FFmpeg, but about the same as…

Why not use a lossless codec if you care about quality? Or use Opus, descent for specht and works pretty much anywhere these days.

Re: FFmpeg 9.1's new AAC encoder

#23

Post by xuhu »

superzazu wrote: post #8

> The encoder was mainly optimized for 48Khz audio. Get over it. It's 2026, resampling is free, 48Khz is the standard. 44.1Khz will work, and so will 96Khz but use 48Khz if you want the best quality. Is 48kHz really the standard nowadays?

For one, audio transcription services that use Whisper will sample the input down to 16Khz mono first.

Re: FFmpeg 9.1's new AAC encoder

#24

Post by zamadatix »

duped wrote: post #19
atoav wrote: post #12

Earlier quoted context omitted.

More or less. Streaming is often done with 48, video content has ben 48 for a while now, so unless you still produce content for CDs it is the standard. 44100 Hz had reasons no longer really needed (storing audio in 3 samples per line in VHS: 490 lines × 3 samples × 30 GPS = 44100 sample/s). Qualitywise both are more than enough snd 99.99% of people would not be able to tell it apart in a blind test. Higher sample ra…

> Higher sample rates than 48kHz only needed when you want to pitch down ultrasonic recordings (of whales, bats and other such animals for example). There are numerous use cases for higher sample rates that go beyond this but it's hard to talk about it without starting flame wars filled with junk science.

Say it or don't but "I have evidence otherwise but don't think I should say" is just as bad a flame war gateway as tempting the junk science audiophiles directly.

Re: FFmpeg 9.1's new AAC encoder

#25

Post by repelsteeltje »

ndiddy wrote: post #20
cogman10 wrote: post #16

Man what a showcase for Opus this is. Don't get me wrong, this sort of thing is a valuable exercise and we are better off with better encoders for these older codecs. But look at the numbers for Opus on this benchmark. It simply blows all the AAC encoders out of the water even at 64 kbps.

The biggest advantage for having a good AAC encoder isn't efficiency, it's that for nearly the past 2 decades the de facto standard for live streamed video has been RTMP with H.264 video and AAC audio. There is basically no support for any other codecs. If you want to send a video stream to Youtube or Twitch, you will be sending H.264 and AAC. If you want an idea of how ubiquitous this is, I just checked in OBS and i…

Sample accurate editing is with AAC is a pain though. Especially if you also have video, because frame rates are usually incompatible.

If you want flexibility without fully transcoding both audio and video, Opus is your friend

Re: FFmpeg 9.1's new AAC encoder

#26

Post by skydhash »

cogman10 wrote: post #16

Man what a showcase for Opus this is. Don't get me wrong, this sort of thing is a valuable exercise and we are better off with better encoders for these older codecs. But look at the numbers for Opus on this benchmark. It simply blows all the AAC encoders out of the water even at 64 kbps.

I would like Opus, but I’m using a subsonic client on iOS and my choice has been Flac (Alac?), MP3, or AAC. Opus wouldn’t play (There are some that supported it, but I didn’t like their UX).

Re: FFmpeg 9.1's new AAC encoder

#27

Post by CharlesW »

ndiddy wrote: post #20
cogman10 wrote: post #16

Man what a showcase for Opus this is. Don't get me wrong, this sort of thing is a valuable exercise and we are better off with better encoders for these older codecs. But look at the numbers for Opus on this benchmark. It simply blows all the AAC encoders out of the water even at 64 kbps.

The biggest advantage for having a good AAC encoder isn't efficiency, it's that for nearly the past 2 decades the de facto standard for live streamed video has been RTMP with H.264 video and AAC audio. There is basically no support for any other codecs. If you want to send a video stream to Youtube or Twitch, you will be sending H.264 and AAC. If you want an idea of how ubiquitous this is, I just checked in OBS and i…

Plus, at 96+ kbps (assuming an Apple-quality AAC-LC encoder) Opus loses its quality advantage. So at higher bitrates, the benefit of choosing Opus is that encoders/decoders are royalty-free.

Re: FFmpeg 9.1's new AAC encoder

#28

Post by jck86 »

cogman10 wrote: post #16

Man what a showcase for Opus this is. Don't get me wrong, this sort of thing is a valuable exercise and we are better off with better encoders for these older codecs. But look at the numbers for Opus on this benchmark. It simply blows all the AAC encoders out of the water even at 64 kbps.

Choosing a lossy audio codec has become such a no brainer. Either use opus and be done with it or if for some reason opus cannot be used then use aac for compatibility with insane high bitrate for good quality without having to do research on what encoder and mode to pick.

Still having a good quality and default aac encoder is great. Though I don't get why it is mainly CBR.

Re: FFmpeg 9.1's new AAC encoder

#29

Post by legdoge »

superzazu wrote: post #8

> The encoder was mainly optimized for 48Khz audio. Get over it. It's 2026, resampling is free, 48Khz is the standard. 44.1Khz will work, and so will 96Khz but use 48Khz if you want the best quality. Is 48kHz really the standard nowadays?

AAC has a strange quirk that the window size is dependent on the sampling rate, thus requiring a complete psychoacoustics reoptimization of all encoder parameters for each sampling rate, since a 20msec window sounds very different than a 60msec window, to human ears.

This was of course fixed in Opus.

Re: FFmpeg 9.1's new AAC encoder

#30

Post by [deleted] »

superzazu wrote: post #8

> The encoder was mainly optimized for 48Khz audio. Get over it. It's 2026, resampling is free, 48Khz is the standard. 44.1Khz will work, and so will 96Khz but use 48Khz if you want the best quality. Is 48kHz really the standard nowadays?

[deleted]
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