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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

#101
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…

I was using FDK AAC encoder, I didn’t know Apple encoder was available for systems other than Apple. Though I have once compared AAC FDK to Apple AAC at 192kbps, and couldn’t tell the difference, while the old FFmpeg AAC encoder fall apart at this bitrate.

It gets installed when you install iTunes. If you don't want to install iTunes, you can pull out the codec installer by opening an old version of the iTunes installer in 7-zip and extracting the MSI. Here's a copy I keep around for whenever I have to do a screen recording on a new computer, it's signed by Apple so you don't have to trust me. https://www.infochunk.com/obs/AppleApplicationSupport64.msi

Re: FFmpeg 9.1's new AAC encoder

#102
post #76
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…

YouTube actually supports H.265 and VP9 ingest, depending on the streaming protocol. I can actually stream 4K@60 H.265 from my Mac Studio with https://developers.google.com/youtube/v3/live/guides/ingesti...

Nice, glad some sites are finally trying to move forward. It looks like they only support H.265 video with AAC audio, so this should still be helpful for people who are streaming H.265. https://developers.google.com/youtube/v3/live/guides/hls-ing...

Re: FFmpeg 9.1's new AAC encoder

#103
post #69
post #34

Earlier quoted context omitted.

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 I think the advantage of lossless audio is for archival: rip once, archive as lossless; then you can reencode your library with the latest and greatest lossy encoders over time, or just use the lossless if your player can manage it, cpu and storage is le…

I wouldn't call archiving a consumer application but I understand the point. Really it gets back to the word: fidelity. Some say it means "truth" but really it's latin for faithful or in the context of audio, perceptually identical (a faithful representation). Even among highly trained and skilled listeners, lossy codecs are faithful and imperceptible.

Re: FFmpeg 9.1's new AAC encoder

#104

Older I get, more it seems it’s possible to ping pong between rewrites for good reasons (ex. here, metric maxes but I find it hard to believe VBR and not-48 kHz are silly things and not worth investing it)

I always encode my AAC with VBR. Why wouldn't you, right? I guess I'll stick to apple or fdkaac for now.

Why do you record AAC?

Re: FFmpeg 9.1's new AAC encoder

#105

I applaud a new/better FFMPEG AAC encoder, but there are two pretty massive caveats that are mentioned in the specifics that need to be called out: - CBR only - Only optimized for 48khz sampling Not being able to do quality-based variable bitrate encoding is a major gap, and since all of the CD audio in the world is at 44.1k sampling, that seems like a huge miss too.

You can use -q:a, for "true" VBR, but its metrics are a few percent (imperceptable, we still win) less. "The benchmarks I posted were done mainly on 44.1Khz. I tuned by ear on 48Khz data though, so some of the windowing/transient logic is tied to 48Khz. It translated to 44.1Khz well enough that I left it as-is, since the timing difference isn't that large."

Re: FFmpeg 9.1's new AAC encoder

#106
post #92
post #6

Earlier quoted context omitted.

It's fixed now. Our software follows redirs and somehow we got a 302 to our own IP. Perhaps it is someone's idea of a bot detector?

Unrelated: Hey, I sent hn@ycombinator.com two emails. One was May 6th, the other June 18th (UTC+2). The former's subject is "Broken prev/next links sometimes". In the latter, I've asked to let me know if it arrived. It didn't bounce so your email server has acknowledged receipt, but based on fast responses to previous emails and someone else mentioning randomly that you responded quickly to theirs in iirc early June,…

Well, they don't read all the emails -or- respond to @dang, @mod, etc.

Your approaches are, ride a comment (as you did, 9 hours, no response, likely didn't read) or lean on a frequent flyer (the only privilege I have is increased ability to [dead] obvious spam not caught by filters - but it gets my emails seen)

I sent them an email in past minute - Good Luck! (YMMV)

Re: FFmpeg 9.1's new AAC encoder

#107

Earlier quoted context omitted.

No one really offloads AAC, apart from Apple. Opus can be decoded on very cheap microcontrollers entirely in software using the reference library.

On a microcontroller doing nothing else sure. But on a phone, a tablet, a laptop, you absolutely want hardware decode to preserve your battery life.

Audio decode is extremely cheap. It's true that a hardware implementation will be more efficient, but really not a whole lot more.

Re: FFmpeg 9.1's new AAC encoder

#108

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.

Took me a second to realize you were talking about the encoder not the model before going into this article

Re: FFmpeg 9.1's new AAC encoder

#109

Earlier quoted context omitted.

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.

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

And even if you could, would the frequencies that all humans lose with age really be all that essential for the enjoyment of music? We are talking about frequencies most instruments won't even produce unless severely abused.

For some reasons in audiophile-land the magic is always in some elusive outer realms and never right there where the important stuff happens. They spend a fortune on speaker cables, while often not giving a second thought on room acoustics beyond the cosmetic. The magic sparkle is all the way in the ultrasonic, while their listening spaces have deep nulls in the mid-range due to comb filtering from reflective surfaces caused by a lack of acoustic treatment.

I love music (enough to have mixed it for a living) and to me it is very clear how the priorities are ordered when it comes to audio fidelity:

1. Room Acoustics

2. Speakers

3. Electronics & Digital

Going from the back: Assuming you don't get the cheapest of the cheapest and don't abuse the gear by making it do things it wasn't build for electronics and digital audio nowadays is transparent. That means, it essentially sounds the same if operated within spec. Even a 0,50 € IC will have distortion figures so staggeringly low it is below human perception and equipment is getting better still. A decent opamp can have distortion figures like 0.005 % THD with a linear frequency response all the way up to radio frequencies. There can be challenges with driving very weird speakers or headphones, but if you hsve the right combination of gear it doesn't have to be expensive to be indistinguishably good in it's audio performance.

This means speakers are way more important thsn the electronics before it. Their distortion numbers are multiple magnitudes higher (in the ball park of 3% THD), their frequency response is inherently problematic (often many dBs up and down even in expensive speakers), they will hsve different beaming characteristics st different frequencies, small speakers lack bass, placement is essential, etc. So getting good speakers is important.

But all of this is dwarfed by the impacts acoustics. The position of the speakers alone makes a huge difference. The impact of an acoustically untreated space is severe: you can get a completely smeared time response with deep nulls of 20dB and more while other frequencies are highly resonant. Even a budget speaker won't have problems of that magnitude.

So get some ok electronics, even more ok speakers, but invest the bulk of the money/time into the setup of the room itself.

Many adiophiles have that priority list reversed. Room acoustics suck. You need to measure a lot, add ugly absorbers in inconvenient places, can't place speakers where they look nice and conserve space, but need to place them where they work well acoustically, there is no ideal solution and everything is a compromise. So buying a gold plated HDMI cable and imagining the improvement appears to be better. Only that you might be doing it in a room where a positional difference of a few centimeters changes the frequency response of the listening position massively.

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