Live data from Hacker News

FFmpeg 9.1's new AAC encoder

hydrogenaudio.org

31–40 of 162 posts

Re: FFmpeg 9.1's new AAC encoder

#31

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).

You might like Poppy (in beta), which supports all media servers (including OpenSubsonic/Navidrome) and Opus as a first-class music format. https://www.reddit.com/r/PoppyApp/comments/1tiyki0/about_pop...

Re: FFmpeg 9.1's new AAC encoder

#32

> 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?

48kHz makes alignment between video and audio so much easier. (I.e.: Lip synchronization after edits)

Re: FFmpeg 9.1's new AAC encoder

#33
post #19

Earlier quoted context omitted.

> 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.

I know that with oscilloscopes, it’s recommended to use 5x instead of nuquist 2x of the highest frequency you want to use., but the most reasonable argument I’ve heard for higher than 48kHz sampling is digital audio effects.

But for the end result 48kHz is more than necessary. I can’t even hear any frequency above 17kHz.

Re: FFmpeg 9.1's new AAC encoder

#34
post #19

Earlier quoted context omitted.

> 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.

Higher sample rates are lower latency for the same block size and resampling is not "free" (pick 2: performance, aliasing, latency) so there can be advantages to working with audio archived at higher sample rates.

But all the advantages come down to professional or editing use cases. There's next to zero advantage to using it as a storage format for listening. Just like 24 bit audio (do you have an amp with 96dB SNR?).

Just personally, I have seen little evidence (personally, professionally, or academically) that there is any advantage for lossless audio for consumer applications. For professional applications there are plenty, and it's endlessly tiring to convince people that "no, actually I need 96kHz for my use case."

Where the audiophiles have _some_ argument here is the design of reconstruction filters which I've heard alleged can perform better in the audible frequency range if the stop band is outside of it. But I have never personally tested this, nor cared enough to. But the theory is sound.

Whether or not it's perceptible depends on what you're measuring, though. In theory, there should be perceptual differences in sound localization if your DAC's reconstruction filter is at 24kHz vs 48kHz since it will change the group delay in a critical frequency region, where you'll get sound at >~2kHz arriving later at the lower sample rate. I think it would be extremely hard to test this though, because humans are really shitty at sound localization to begin with, and practically speaking most recorded material is processed to shit in that frequency range to intentionally decorrelate the channels for the perception of "width."

Re: FFmpeg 9.1's new AAC encoder

#36
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.

> Why not use a lossless codec if you care about quality?

(1) Lossy codecs are transparent at half the file size (or less) of FLAC/ALAC.

(2) AAC (strictly, AAC-LC) is universal, where FLAC and Opus are not yet there.

Re: FFmpeg 9.1's new AAC encoder

#37
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.

You can care about quality to the extent that a lossy codec allows. Lossless is not always necessary or wanted. This is like saying “why care about transcoding quality when you can keep the video as is?”. There’s a myriad of use cases and preferences at play here.

Re: FFmpeg 9.1's new AAC encoder

#38

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 think the biggest issue with Opus is the problem with its specification being lacking, see:

https://nothings.org/stb/stb_opus.html

This essentially causes opus to never be used in games or in things in stores that may have issues with specific licenses.

Re: FFmpeg 9.1's new AAC encoder

#39
post #20

Earlier quoted context omitted.

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

Opus is your friend as long as the software you’re using supports it—besides, Apple’s AAC-LC can beat out Opus in low bitrates scenarios.

Whether you like it or not, AAC is still the standard.

Re: FFmpeg 9.1's new AAC encoder

#40

> 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?

I think the closest thing to an actual "standard" is AES5-2018, "Recommended practice for professional digital audio".

Abstract:

> A sampling frequency of 48 kHz is recommended for the origination, processing, and interchange of audio programs employing pulse-code modulation. Recognition is also given to the use of a 44.1-kHz sampling frequency related to certain consumer digital applications, the use of a 32-kHz sampling frequency for transmission-related applications, and the use of a 96-kHz sampling frequency for applications requiring a higher bandwidth or more relaxed anti-alias filtering. This revision further quantifies the preferred choices for higher sampling frequencies.

Edit: From my personal perspective, 44.1kHz is a legacy minor annoyance

Post reply on HN