FFmpeg 9.1's new AAC encoder(hydrogenaudio.org) |
FFmpeg 9.1's new AAC encoder(hydrogenaudio.org) |
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.
If you want flexibility without fully transcoding both audio and video, Opus is your friend
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.
The licensing restriction is unfortunate, but only restrictive for those with very specific goals, under normal conditions BSD is a wonderful license for game devs since you’re free to use the code and only have to add an acknowledgement somewhere.
I suppose a public domain game might hit the same limitation, though as a non-lawyer I would guess the chance of anyone with standing trying to sue anyone implementing from this spec is realistically zero (though I don’t fault stb for being unwilling to roll those dice!)
https://www.audiokinetic.com/en/community/blog/a-guide-for-c...
Still having a good quality and default aac encoder is great. Though I don't get why it is mainly CBR.
Falser words hath never been spoken.
It also uses less battery on my Rockbox'd Clip+.
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.
Is 48kHz really the standard nowadays?
I don't know what PNS is, but I bet this has been bothering someone's niche use-case for 20 years
Link should be: https://hydrogenaudio.org/index.php/topic,129691.0.html
So maybe Core Audio will continue to be the best when TVBR is available, but I'm hopeful the new FFmpeg encoder will be "good enough", especially if more folks find and contribute problem samples to help tune it.
(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.
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
This was of course fixed in Opus.
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 rates than 48kHz only needed when you want to pitch down ultrasonic recordings (of whales, bats and other such animals for example).
Aside from this higher than 48 kHz sample rates may have only downsides, like increased size and potential distortion in the ultrasonic frequency range that has sidebands in the audible range. Yet there is a persistent, but unscientific "more-is-better"-crowd in the HiFi-sector.
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.
44.1kHz, isn't that what lameMP3 uses as default?
Whether you like it or not, AAC is still the standard.
But would you really store new material encoded in Vorbis just to be able to play it on an old device? Vorbis can sound fine, even at lower bitrates like 128k or 96k, but Opus would sound much better. So perhaps then use Vorbis at higher bitrates like +192k? I prefer Vorbis to Aac but at that bitrate minor intricacies of the container format become more important than the codec because audio quality wise they are near indistinguishable.
Our software follows redirs and somehow we got a 302 to our own IP. Perhaps it is someone's idea of a bot detector?
* quick email to HN@ycombinator.com with a "Help Me please!! and link ( mods can edit link in and sideline (hide) these comments )
* Just live with the rotting fish head of public boo boo (we've all made mistakes, as the Dalek said whilst climbing down off the dustbin)
* I can kill the whole thing dead.
Why? Care to explain?
I have fully switched to opus for lossy since I cannot be bothered to find the sweet spot for aac encoders and bitrate. Opus simply is too good and convenient and has been for ages.
What other lossy codec is better and for what reason? Under what circumstances and use cases? I really need put effort in looking for edge cases to not choose opus.
Aac is good too, but way too many choices to make for storing mass material for the long term and be sure the quality is always good enough.
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."
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 less of a limiting factor for players than 20 years ago.
I don't know how many people are actually managing their libraries these days though, so I dunno if makes a huge difference.
This a truly bizarre statement. On the one hand, of course higher sampling rates are lower latency for the same block size measured in samples. But all sampling rates have (almost [0]) identical latency for the same block size measured in time and lower sampling rates allow less computation for those shorter blocks.
[0] If you are concerned about needing to know future samples in order to calculate the actual signal amplitude at a time between samples, then (a) this matters less at higher sampling rates and (b) this is at most a small number of samples and we're talking about block sizes that presumably exceed, say, 5, so this isn't really a big deal.
And if your goal is latency, it makes far more sense to change the block size rather than the sample rate.
> But all the advantages come down to professional or editing use cases.
That sounds about right.
But for the end result 48kHz is more than necessary. I can’t even hear any frequency above 17kHz.